Real-Time Streaming Protocol Version 2.0Columbia University1214 Amsterdam AvenueNew YorkNY10027United States of Americaschulzrinne@cs.columbia.eduCiscoUnited States of Americaanrao@cisco.comSan FranciscoCAUnited States of Americarobla@robla.netEricssonFaeroegatan 2StockholmSE-164 80Swedenmagnus.westerlund@ericsson.comUniversity of Applied Sciences DarmstadtHaardtring 10064295 DarmstadtGermanymls.ietf@gmail.comhttp://www.stiemerling.org
Real-time Applications and Infrastructure Area
MMUSIC Working Groupmmusic, RTSP, RTSP/2.0, real-time streaming protocolThis memorandum defines the Real-Time Streaming Protocol (RTSP)
version 2.0, which obsoletes RTSP version 1.0 defined in RFC 2326.RTSP is an application-layer protocol for the setup and control of
the delivery of data with real-time properties. RTSP provides an
extensible framework to enable controlled, on-demand delivery of
real-time data, such as audio and video. Sources of data can include
both live data feeds and stored clips. This protocol is intended to
control multiple data delivery sessions; provide a means for choosing
delivery channels such as UDP, multicast UDP, and TCP; and provide a
means for choosing delivery mechanisms based upon RTP (RFC 3550).This memo defines version 2.0 of the Real-Time Streaming Protocol
(RTSP 2.0). RTSP 2.0 is an application-layer protocol for the setup and
control over the delivery of data with real-time properties, typically
streaming media. Streaming media is, for instance, video on demand or
audio live streaming. Put simply, RTSP acts as a "network remote
control" for multimedia servers.The protocol operates between RTSP 2.0 clients and servers, but it
also supports the use of proxies placed between clients and servers.
Clients can request information about streaming media from servers by
asking for a description of the media or use media description provided
externally. The media delivery protocol is used to establish the media
streams described by the media description. Clients can then request to
play out the media, pause it, or stop it completely. The requested media
can consist of multiple audio and video streams that are delivered as
time-synchronized streams from servers to clients.RTSP 2.0 is a replacement of RTSP 1.0
and this document obsoletes that specification. This protocol is based
on RTSP 1.0 but is not backwards compatible other than in the basic
version negotiation mechanism. The changes between the two documents are
listed in . There are many reasons why RTSP
2.0 can't be backwards compatible with RTSP 1.0; some of the main ones
are as follows:Most headers that needed to be extensible did not define the
allowed syntax, preventing safe deployment of extensions;the changed behavior of the PLAY method when received in Play
state;the changed behavior of the extensibility model and its
mechanism; andthe change of syntax for some headers.There are so many small updates that changing versions became
necessary to enable clarification and consistent behavior. Anyone
implementing RTSP for a new use case in which they have not installed
RTSP 1.0 should only implement RTSP 2.0 to avoid having to deal with
RTSP 1.0 inconsistencies.This document is structured as follows. It begins with an overview of
the protocol operations and its functions in an informal way. Then, a
set of definitions of terms used and document conventions is introduced.
These are followed by the actual RTSP 2.0 core protocol specification.
The appendices describe and define some functionalities that are not
part of the core RTSP specification, but which are still important to
enable some usages. Among them, the RTP usage is defined in , the Session Description Protocol (SDP) usage
with RTSP is defined in , and the
"text/parameters" file format , are
three normative specification appendices. Other appendices include a
number of informational parts discussing the changes, use cases,
different considerations or motivations.This section provides an informative overview of the different
mechanisms in the RTSP 2.0 protocol to give the reader a high-level
understanding before getting into all the specific details. In case of
conflict with this description and the later sections, the later
sections take precedence. For more information about use cases
considered for RTSP, see .RTSP 2.0 is a bidirectional request and response protocol that first
establishes a context including content resources (the media) and then
controls the delivery of these content resources from the provider to
the consumer. RTSP has three fundamental parts: Session Establishment,
Media Delivery Control, and an extensibility model described below. The
protocol is based on some assumptions about existing functionality to
provide a complete solution for client-controlled real-time media
delivery.RTSP uses text-based messages, requests and responses, that may
contain a binary message body. An RTSP request starts with a method line
that identifies the method, the protocol, and version and the resource
on which to act. The resource is identified by a URI and the hostname
part of the URI is used by RTSP client to resolve the IPv4 or IPv6
address of the RTSP server. Following the method line are a number of
RTSP headers. These lines are ended by two consecutive carriage return
line feed (CRLF) character pairs. The message body, if present, follows
the two CRLF character pairs, and the body's length is described by a
message header. RTSP responses are similar, but they start with a
response line with the protocol and version followed by a status code
and a reason phrase. RTSP messages are sent over a reliable transport
protocol between the client and server. RTSP 2.0 requires clients and
servers to implement TCP and TLS over TCP as mandatory transports for
RTSP messages.RTSP exists to provide access to multimedia presentations and
content but tries to be agnostic about the media type or the actual
media delivery protocol that is used. To enable a client to implement
a complete system, an RTSP-external mechanism for describing the
presentation and the delivery protocol(s) is used. RTSP assumes that
this description is either delivered completely out of band or as a
data object in the response to a client's request using the DESCRIBE method.Parameters that commonly have to be included in the presentation
description are the following:The number of media streams;the resource identifier for each media stream/resource that is
to be controlled by RTSP;the protocol that will be used to deliver each media
stream;the transport protocol parameters that are not negotiated or
vary with each client;the media-encoding information enabling a client to correctly
decode the media upon reception; andan aggregate control resource identifier.RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference
media resources and aggregates under common control (see ).This specification describes in how
one uses SDP for describing the
presentation.The RTSP client can request the establishment of an RTSP session
after having used the presentation description to determine which
media streams are available, which media delivery protocol is used,
and the resource identifiers of the media streams. The RTSP session is
a common context between the client and the server that consists of
one or more media resources that are to be under common media delivery
control.The client creates an RTSP session by sending a request using the
SETUP method to the server. In the
Transport header of the SETUP
request, the client also includes all the transport parameters
necessary to enable the media delivery protocol to function. This
includes parameters that are preestablished by the presentation
description but necessary for any middlebox to correctly handle the
media delivery protocols. The Transport header in a request may
contain multiple alternatives for media delivery in a prioritized
list, which the server can select from. These alternatives are
typically based on information in the presentation description.When receiving a SETUP request, the server determines if the media
resource is available and if one or more of the of the transport
parameter specifications are acceptable. If that is successful, an
RTSP session context is created and the relevant parameters and state
is stored. An identifier is created for the RTSP session and included
in the response in the Session
header. The SETUP response includes a Transport header that
specifies which of the alternatives has been selected and relevant
parameters.A SETUP request that references an existing RTSP session but
identifies a new media resource is a request to add that media
resource under common control with the already-present media resources
in an aggregated session. A client can expect this to work for all
media resources under RTSP control within a multimedia content
container. However, a server will likely refuse to aggregate resources
from different content containers. Even if an RTSP session contains
only a single media stream, the RTSP session can be referenced by the
aggregate control URI.To avoid an extra round trip in the session establishment of
aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e.,
the client can send multiple requests back-to-back without waiting
first for the completion of any of them. The client uses a
client-selected identifier in the Pipelined-Requests header to
instruct the server to bind multiple requests together as if they
included the session identifier.The SETUP response also provides additional information about the
established sessions in a couple of different headers. The Media-Properties header includes
a number of properties that apply for the aggregate that is valuable
when doing media delivery control and configuring user interface. The
Accept-Ranges header informs
the client about range formats that the server supports for these
media resources. The Media-Range
header informs the client about the time range of the media
currently available.After having established an RTSP session, the client can start
controlling the media delivery. The basic operations are "begin
playback", using the PLAY method and
"suspend (pause) playback" by using the PAUSE
method. PLAY also allows for choosing the starting media
position from which the server should deliver the media. The
positioning is done by using the Range
header that supports several different time formats: Normal Play Time (NPT), Society of Motion Picture and Television Engineers
(SMPTE) Timestamps, and absolute
time. The Range header also allows the client to specify a
position where delivery should end, thus allowing a specific interval
to be delivered.The support for positioning/searching within media content depends
on the content's media properties. Content exists in a number of
different types, such as on-demand, live, and live with simultaneous
recording. Even within these categories, there are differences in how
the content is generated and distributed, which affect how it can be
accessed for playback. The properties applicable for the RTSP session
are provided by the server in the SETUP response using the Media-Properties header. These
are expressed using one or several independent attributes. A first
attribute is Random-Access, which indicates whether positioning is
possible, and with what granularity. Another aspect is whether the
content will change during the lifetime of the session. While
on-demand content will be provided in full from the beginning, a live
stream being recorded results in the length of the accessible content
growing as the session goes on. There also exists content that is
dynamically built by a protocol other than RTSP and, thus, also
changes in steps during the session, but maybe not continuously.
Furthermore, when content is recorded, there are cases where the
complete content is not maintained, but, for example, only the last
hour. All of these properties result in the need for mechanisms that
will be discussed below.When the client accesses on-demand content that allows random
access, the client can issue the PLAY request for any point in the
content between the start and the end. The server will deliver media
from the closest random access point prior to the requested point and
indicate that in its PLAY response. If the client issues a PAUSE, the
delivery will be halted and the point at which the server stopped will
be reported back in the response. The client can later resume by
sending a PLAY request without a Range header. When the server is
about to complete the PLAY request by delivering the end of the
content or the requested range, the server will send a PLAY_NOTIFY request indicating
this.When playing live content with no extra functions, such as
recording, the client will receive the live media from the server
after having sent a PLAY request. Seeking in such content is not
possible as the server does not store it, but only forwards it from
the source of the session. Thus, delivery continues until the client
sends a PAUSE request, tears down the session, or the content
ends.For live sessions that are being recorded, the client will need to
keep track of how the recording progresses. Upon session
establishment, the client will learn the current duration of the
recording from the Media-Range header. Because the recording is
ongoing, the content grows in direct relation to the time passed.
Therefore, each server's response to a PLAY request will contain the
current Media-Range header. The server should also regularly send
(approximately every 5 minutes) the current media range in a
PLAY_NOTIFY request (). If the live
transmission ends, the server must send a PLAY_NOTIFY request with the
updated Media-Properties indicating that the content stopped being a
recorded live session and instead became on-demand content; the
request also contains the final media range. While the live delivery
continues, the client can request to play the current live point by
using the NPT timescale symbol "now", or it can request a specific
point in the available content by an explicit range request for that
point. If the requested point is outside of the available interval,
the server will adjust the position to the closest available point,
i.e., either at the beginning or the end.A special case of recording is that where the recording is not
retained longer than a specific time period; thus, as the live
delivery continues, the client can access any media within a moving
window that covers, for example, "now" to "now" minus 1 hour. A client
that pauses on a specific point within the content may not be able to
retrieve the content anymore. If the client waits too long before
resuming the pause point, the content may no longer be available. In
this case, the pause point will be adjusted to the closest point in
the available media.A session may have additional state or functionality that affects
how the server or client treats the session or content, how it
functions, or feedback on how well the session works. Such extensions
are not defined in this specification, but they may be covered in
various extensions. RTSP has two methods for retrieving and setting
parameter values on either the client or the server: GET_PARAMETER and SET_PARAMETER. These methods carry
the parameters in a message body of the appropriate format. One can
also use headers to query state with the GET_PARAMETER method. As an
example, clients needing to know the current media range for a
time-progressing session can use the GET_PARAMETER method and include
the media range. Furthermore, synchronization information can be
requested by using a combination of RTP-Info and Range.RTSP 2.0 does not have a strong mechanism for negotiating the
headers or parameters and their formats. However, responses will
indicate request-headers or parameters that are not supported. A
priori determination of what features are available needs to be done
through out-of-band mechanisms, like the session description, or
through the usage of feature
tags.This document specifies how media is delivered with RTP over UDP,
TCP, or the RTSP connection. Additional
protocols may be specified in the future as needed.The usage of RTP as a media delivery protocol requires some
additional information to function well. The PLAY response contains
information to enable reliable and timely delivery of how a client
should synchronize different sources in the different RTP sessions. It
also provides a mapping between RTP timestamps and the content-time
scale. When the server wants to notify the client about the completion
of the media delivery, it sends a PLAY_NOTIFY request to the client.
The PLAY_NOTIFY request includes information about the stream end,
including the last RTP sequence number for each stream, thus enabling
the client to empty the buffer smoothly.The basic playback functionality of RTSP enables delivery of a
range of requested content to the client at the pace intended by the
content's creator. However, RTSP can also manipulate the delivery to
the client in two ways.The ratio of media-content time delivered
per unit of playback time.The ratio of playback time delivered per
unit of wallclock time.Both affect the media delivery per time unit. However, they
manipulate two independent timescales and the effects are possible
to combine.Scale is used for fast-forward or
slow-motion control as it changes the amount of content timescale
that should be played back per time unit. Scale > 1.0, means fast
forward, e.g., scale = 2.0 results in that 2 seconds of content
being played back every second of playback. Scale = 1.0 is the
default value that is used if no scale is specified, i.e., playback
at the content's original rate. Scale values between 0 and 1.0
provide for slow motion. Scale can be negative to allow for reverse
playback in either regular pace (scale = -1.0), fast backwards
(scale < -1.0), or slow-motion backwards (-1.0 < scale <
0). Scale = 0 would be equal to pause and is not allowed.In most cases, the realization of scale means server-side
manipulation of the media to ensure that the client can actually
play it back. The nature of these media manipulations and when they
are needed is highly media-type dependent. Let's consider two common
media types, audio and video.It is very difficult to modify the playback rate of audio.
Typically, no more than a factor of two is possible while
maintaining intelligibility by changing the pitch and rate of
speech. Music goes out of tune if one tries to manipulate the
playback rate by resampling it. This is a well-known problem, and
audio is commonly muted or played back in short segments with skips
to keep up with the current playback point.For video, it is possible to manipulate the frame rate, although
the rendering capabilities are often limited to certain frame rates.
Also, the allowed bitrates in decoding, the structure used in the
encoding, and the dependency between frames and other capabilities
of the rendering device limits the possible manipulations.
Therefore, the basic fast-forward capabilities often are implemented
by selecting certain subsets of frames.Due to the media restrictions, the possible scale values are
commonly restricted to the set of realizable scale ratios. To enable
the clients to select from the possible scale values, RTSP can
signal the supported scale ratios for the content. To support
aggregated or dynamic content, where this may change during the
ongoing session and dependent on the location within the content, a
mechanism for updating the media properties and the scale factor
currently in use, exists.Speed affects how much of the
playback timeline is delivered in a given wallclock period. The
default is Speed = 1 which means to deliver at the same rate the
media is consumed. Speed > 1 means that the receiver will get
content faster than it regularly would consume it. Speed < 1
means that delivery is slower than the regular media rate. Speed
values of 0 or lower have no meaning and are not allowed. This
mechanism enables two general functionalities. One is client-side
scale operations, i.e., the client receives all the frames and makes
the adjustment to the playback locally. The second is delivery
control for the buffering of media. By specifying a speed over 1.0,
the client can build up the amount of playback time it has present
in its buffers to a level that is sufficient for its needs.A naive implementation of Speed would only affect the
transmission schedule of the media and has a clear impact on the
needed bandwidth. This would result in the data rate being
proportional to the speed factor. Speed = 1.5, i.e., 50% faster than
normal delivery, would result in a 50% increase in the
data-transport rate. Whether or not that can be supported depends
solely on the underlying network path. Scale may also have some
impact on the required bandwidth due to the manipulation of the
content in the new playback schedule. An example is fast forward
where only the independently decodable intra-frames are included in
the media stream. This usage of solely intra-frames increases the
data rate significantly compared to a normal sequence with the same
number of frames, where most frames are encoded using
prediction.This potential increase of the data rate needs to be handled by
the media sender. The client has requested that the media be
delivered in a specific way, which should be honored. However, the
media sender cannot ignore if the network path between the sender
and the receiver can't handle the resulting media stream. In that
case, the media stream needs to be adapted to fit the available
resources of the path. This can result in a reduced media
quality.The need for bitrate adaptation becomes especially problematic in
connection with the Speed semantics. If the goal is to fill up the
buffer, the client may not want to do that at the cost of reduced
quality. If the client wants to make local playout changes, then it
may actually require that the requested speed be honored. To resolve
this issue, Speed uses a range so that both cases can be supported.
The server is requested to use the highest possible speed value
within the range, which is compatible with the available bandwidth.
As long as the server can maintain a speed value within the range,
it shall not change the media quality, but instead modify the actual
delivery rate in response to available bandwidth and reflect this in
the Speed value in the response. However, if this is not possible,
the server should instead modify the media quality to respect the
lowest speed value and the available bandwidth.This functionality enables the local scaling implementation to
use a tight range, or even a range where the lower bound equals the
upper bound, to identify that it requires the server to deliver the
requested amount of media time per delivery time, independent of how
much it needs to adapt the media quality to fit within the available
path bandwidth. For buffer filling, it is suitable to use a range
with a reasonable span and with a lower bound at the nominal media
rate 1.0, such as 1.0 - 2.5. If the client wants to reduce the
buffer, it can specify an upper bound that is below 1.0 to force the
server to deliver slower than the nominal media rate.The session context that has been established is kept alive by
having the client show liveness. This is done in two main ways:Media-transport protocol keep-alive. RTP Control Protocol
(RTCP) may be used when using RTP.Any RTSP request referencing the session context. discusses the methods for showing
liveness in more depth. If the client fails to show liveness for more
than the established session timeout value (normally 60 seconds), the
server may terminate the context. Other values may be selected by the
server through the inclusion of the timeout parameter in the session
header.The session context is normally terminated by the client sending a
TEARDOWN request to the server
referencing the aggregated control URI. An individual media resource
can be removed from a session context by a TEARDOWN request
referencing that particular media resource. If all media resources are
removed from a session context, the session context is terminated.A client may keep the session alive indefinitely if allowed by the
server; however, a client is advised to release the session context
when an extended period of time without media delivery activity has
passed. The client can re-establish the session context if required
later. What constitutes an extended period of time is dependent on the
client, server, and their usage. It is recommended that the client
terminate the session before ten times the session timeout value has
passed. A server may terminate the session after one session timeout
period without any client activity beyond keep-alive. When a server
terminates the session context, it does so by sending a TEARDOWN
request indicating the reason.A server can also request that the client tear down the session and
re-establish it at an alternative server, as may be needed for
maintenance. This is done by using the REDIRECT method. The Terminate-Reason header is used
to indicate when and why. The Location header indicates where it
should connect if there is an alternative server available. When the
deadline expires, the server simply stops providing the service. To
achieve a clean closure, the client needs to initiate session
termination prior to the deadline. In case the server has no other
server to redirect to, and it wants to close the session for
maintenance, it shall use the TEARDOWN method with a Terminate-Reason
header.RTSP is quite a versatile protocol that supports extensions in many
different directions. Even this core specification contains several
blocks of functionality that are optional to implement. The use case
and need for the protocol deployment should determine what parts are
implemented. Allowing for extensions makes it possible for RTSP to
address additional use cases. However, extensions will affect the
interoperability of the protocol; therefore, it is important that they
can be added in a structured way.The client can learn the capability of a server by using the OPTIONS method and the Supported header. It can also try and
possibly fail using new methods or require that particular features be
supported using the Require or Proxy-Require header.The RTSP, in itself, can be extended in three ways, listed here in
increasing order of the magnitude of changes supported: Existing methods can be extended with new parameters, for
example, headers, as long as these parameters can be safely
ignored by the recipient. If the client needs negative
acknowledgment when a method extension is not supported, a tag
corresponding to the extension may be added in the field of the
Require or Proxy-Require headers.New methods can be added. If the recipient of the message does
not understand the request, it must respond with error code 501
(Not Implemented) so that the sender can avoid using this method
again. A client may also use the OPTIONS method to inquire about
methods supported by the server. The server must list the methods
it supports using the Public response-header.A new version of the protocol can be defined, allowing almost
all aspects (except the position of the protocol version number)
to change. A new version of the protocol must be registered
through a Standards Track document.The basic capability discovery mechanism can be used to both
discover support for a certain feature and to ensure that a feature is
available when performing a request. For a detailed explanation of
this, see .New media delivery protocols may be added and negotiated at session
establishment, in addition to extensions to the core protocol. Certain
types of protocol manipulations can be done through parameter formats
using SET_PARAMETER and GET_PARAMETER.All the mechanisms specified in this document are described in both
prose and the Augmented Backus-Naur form (ABNF) described in detail in
.Indented paragraphs are used to provide informative background and
motivation. This is intended to give readers who were not involved
with the formulation of the specification an understanding of why
things are the way they are in RTSP.The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
.The word, "unspecified" is used to indicate functionality or
features that are not defined in this specification. Such
functionality cannot be used in a standardized manner without further
definition in an extension specification to RTSP.The concept of controlling
multiple streams using a single timeline, generally one maintained
by the server. A client, for example, uses aggregate control when
it issues a single play or pause message to simultaneously control
both the audio and video in a movie. A session that is under
aggregate control is referred to as an "aggregated session".The URI used in an RTSP
request to refer to and control an aggregated session. It
normally, but not always, corresponds to the presentation URI
specified in the session description. See for more information.The client is the requester of media service
from the media server.A transport-layer virtual circuit
established between two programs for the purpose of
communication.A file that may contain multiple
media streams that often constitute a presentation when played
together. The concept of a container file is not embedded in the
protocol. However, RTSP servers may offer aggregate control on the
media streams within these files.Data where there is a timing
relationship between source and sink; that is, the sink needs to
reproduce the timing relationship that existed at the source. The
most common examples of continuous media are audio and motion
video. Continuous media can be real time (interactive or
conversational), where there is a "tight" timing relationship
between source and sink or it can be streaming where the
relationship is less strict.A tag representing a certain set of
functionality, i.e., a feature.An Internationalized Resource Identifier is
similar to a URI but allows characters from the whole Universal
Character Set (Unicode/ISO 10646), rather than the US-ASCII only.
See for more information.A live presentation or session originates
media from an event taking place at the same time as the media
delivery. Live sessions often have an unbound or only loosely
defined duration and seek operations may not be possible.The datatype- or
codec-specific initialization. This includes such things as clock
rates, color tables, etc. Any transport-independent information
that is required by a client for playback of a media stream occurs
in the media initialization phase of stream setup.A parameter specific to a media
type that may be changed before or during stream delivery.The server providing media-delivery
services for one or more media streams. Different media streams
within a presentation may originate from different media servers.
A media server may reside on the same host or on a different host
from which the presentation is invoked.A single media instance, e.g., an
audio stream or a video stream as well as a single whiteboard or
shared application group. When using RTP, a stream consists of all
RTP and RTCP packets created by a media source within an RTP
session.The basic unit of RTSP communication,
consisting of a structured sequence of octets matching the syntax
defined in and transmitted over a
transport between RTSP agents. A message is either a request or a
response.The information transferred as the
payload of a message (request or response). A message body
consists of meta-information in the form of message body headers
and content in the form of an arbitrary number of data octets, as
described in .Control of a single media
stream.A set of one or more streams presented
to the client as a complete media feed and described by a
presentation description as defined below. Presentations with more
than one media stream are often handled in RTSP under aggregate
control.A presentation description
contains information about one or more media streams within a
presentation, such as the set of encodings, network addresses, and
information about the content. Other IETF protocols, such as SDP
(), use the term "session" for a
presentation. The presentation description may take several
different formats, including but not limited to SDP format.An RTSP response to a request. One type of
RTSP message. If an HTTP response is meant, it is indicated
explicitly.An RTSP request. One type of RTSP message.
If an HTTP request is meant, it is indicated explicitly.The URI used in a request to indicate
the resource on which the request is to be performed.Either an RTSP client, an RTSP server,
or an RTSP proxy. In this specification, there are many
capabilities that are common to these three entities such as the
capability to send requests or receive responses. This term will
be used when describing functionality that is applicable to all
three of these entities.A stateful abstraction upon which the
main control methods of RTSP operate. An RTSP session is a common
context; it is created and maintained on a client's request and
can be destroyed by either the client or server. It is established
by an RTSP server upon the completion of a successful SETUP
request (when a 200 OK response is sent) and is labeled with a
session identifier at that time. The session exists until timed
out by the server or explicitly removed by a TEARDOWN request. An
RTSP session is a stateful entity; an RTSP server maintains an
explicit session state machine (see )
where most state transitions are triggered by client requests. The
existence of a session implies the existence of state about the
session's media streams and their respective transport mechanisms.
A given session can have one or more media streams associated with
it. An RTSP server uses the session to aggregate control over
multiple media streams.The server on which a given resource
resides.Requesting playback from a particular point
in the content time line.The negotiation of
transport information (e.g., port numbers, transport protocols)
between the client and the server.A Universal Resource Identifier; see . The URIs used in RTSP are generally URLs as
they give a location for the resource. As URLs are a subset of
URIs, they will be referred to as URIs to cover also the cases
when an RTSP URI would not be a URL.A Universal Resource Locator is a URI that
identifies the resource through its primary access mechanism
rather than identifying the resource by name or by some other
attribute(s) of that resource.This specification defines version 2.0 of RTSP.RTSP uses a "<major>.<minor>" numbering scheme to
indicate versions of the protocol. The protocol versioning policy is
intended to allow the sender to indicate the format of a message and
its capacity for understanding further RTSP communication rather than
the features obtained via that communication. No change is made to the
version number for the addition of message components that do not
affect communication behavior or that only add to extensible field
values.The <minor> number is incremented when the changes made to
the protocol add features that do not change the general message
parsing algorithm but that may add to the message semantics and imply
additional capabilities of the sender. The <major> number is
incremented when the format of a message within the protocol is
changed. The version of an RTSP message is indicated by an
RTSP-Version field in the first line of the message. Note that the
major and minor numbers MUST be treated as separate integers and that
each MAY be incremented higher than a single digit. Thus, RTSP/2.4 is
a lower version than RTSP/2.13, which, in turn, is lower than
RTSP/12.3. Leading zeros SHALL NOT be sent and MUST be ignored by
recipients.RTSP 2.0 defines and registers or updates three URI schemes "rtsp",
"rtsps", and "rtspu". The usage of the last, "rtspu", is unspecified
in RTSP 2.0 and is defined here to register the URI scheme that was
defined in RTSP 1.0. The "rtspu" scheme indicates unspecified
transport of the RTSP messages over unreliable transport means (UDP in
RTSP 1.0). An RTSP server MUST respond with an error code indicating
the "rtspu" scheme is not implemented (501) to a request that carries
a "rtspu" URI scheme.The details of the syntax of "rtsp" and "rtsps" URIs have been
changed from RTSP 1.0. These changes include the addition of:Support for an IPv6 literal in the host part and future IP
literals through a mechanism defined in .A new relative format to use in the RTSP elements that is not
required to start with "/".Neither should have any significant impact on
interoperability. If IPv6 literals are needed in the RTSP URI, then
that RTSP server must be IPv6 capable, and RTSP 1.0 is not a fully
IPv6 capable protocol. If an RTSP 1.0 client attempts to process the
URI, the URI will not match the allowed syntax, it will be considered
invalid, and processing will be stopped. This is clearly a failure to
reach the resource; however, it is not a signification issue as RTSP
2.0 support was needed anyway in both server and client. Thus, failure
will only occur in a later step when there is an RTSP version mismatch
between client and server. The second change will only occur inside
RTSP message headers, as the Request-URI must be an absolute URI.
Thus, such usages will only occur after an agent has accepted and
started processing RTSP 2.0 messages, and an agent using RTSP 1.0 only
will not be required to parse such types of relative URIs.This specification also defines the format of RTSP IRIs that can be used as RTSP resource identifiers and
locators on web pages, user interfaces, on paper, etc. However, the
RTSP request message format only allows usage of the absolute URI
format. The RTSP IRI format MUST use the rules and transformation for
IRIs to URIs, as defined in . This allows a
URI that matches the RTSP 2.0 specification, and so is suitable for
use in a request, to be created from an RTSP IRI.The RTSP IRI and URI are both syntax restricted compared to the
generic syntax defined in and : An absolute URI requires the authority part; i.e., a host
identity MUST be provided.Parameters in the path element are prefixed with the reserved
separator ";".The "scheme" and "host" parts of all URIs and IRIs
are case insensitive. All other parts of RTSP URIs and IRIs are case
sensitive, and they MUST NOT be case mapped.The fragment identifier is used as defined in Sections 3.5 and 4.3
of , i.e., the fragment is to be stripped from
the IRI by the requester and not included in the Request-URI. The user
agent needs to interpret the value of the fragment based on the media
type the request relates to; i.e., the media type indicated in
Content-Type header in the response to a DESCRIBE request.The syntax of any URI query string is unspecified and responder
(usually the server) specific. The query is, from the requester's
perspective, an opaque string and needs to be handled as such. Please
note that relative URIs with queries are difficult to handle due to
the relative URI handling rules of RFC 3986. Any change of the path
element using a relative URI results in the stripping of the query,
which means the relative part needs to contain the query.The URI scheme "rtsp" requires that commands be issued via a
reliable protocol (within the Internet, TCP), while the scheme "rtsps"
identifies a reliable transport using secure transport (TLS ); see .For the scheme "rtsp", if no port number is provided in the
authority part of the URI, the port number 554 MUST be used. For the
scheme "rtsps", if no port number is provided in the authority part of
the URI port number, the TCP port 322 MUST be used.A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions of URIs
. URIs may refer to a stream or an aggregate
of streams; i.e., a presentation. Accordingly, requests described in
can apply to either the whole
presentation or an individual stream within the presentation. Note
that some request methods can only be applied to streams, not
presentations, and vice versa.For example, the RTSP URI: rtsp://media.example.com:554/twister/audiotrack may identify the audio stream within the presentation
"twister", which can be controlled via RTSP requests issued over a TCP
connection to port 554 of host media.example.com.Also, the RTSP URI: rtsp://media.example.com:554/twister identifies the presentation "twister", which may be composed
of audio and video streams, but could also be something else, such as
a random media redirector.This does not imply a standard way to reference streams in
URIs. The presentation description defines the hierarchical
relationships in the presentation and the URIs for the individual
streams. A presentation description may name a stream "a.mov" and
the whole presentation "b.mov".The path components of the RTSP URI are opaque to the client and do
not imply any particular file system structure for the server.This decoupling also allows presentation descriptions to be
used with non-RTSP media control protocols simply by replacing the
scheme in the URI.Session identifiers are strings of a length between 8-128
characters. A session identifier MUST be generated using methods that
make it cryptographically random (see ). It is
RECOMMENDED that a session identifier contain 128 bits of entropy,
i.e., approximately 22 characters from a high-quality generator (see
). However, note that the session
identifier does not provide any security against session hijacking
unless it is kept confidential by the client, server, and trusted
proxies.RTSP currently supports three different media-time formats defined
below. Additional time formats may be specified in the future. These
time formats can be used with the Range
header to request playback and specify at which media position
protocol requests actually will or have taken place. They are also
used in description of the media's properties using the Media-Range header. The unqualified
format identifier is used on its own in Accept-Ranges header to declare
supported time formats and also in the Range
header to request the time format used in the response.A timestamp may use a format derived from a Society of Motion
Picture and Television Engineers (SMPTE) specification and expresses
time offsets anchored at the start of the media clip. Relative
timestamps are expressed as SMPTE time
codes for frame-level access accuracy. The time code has the
format:hours:minutes:seconds:frames.subframes with the origin at the start of the clip. The default
SMPTE format is "SMPTE 30 drop" format, with a frame rate of 29.97
frames per second. Other SMPTE codes MAY be supported (such as
"SMPTE 25") through the use of "smpte-type". For SMPTE 30, the
"frames" field in the time value can assume the values 0 through 29.
The difference between 30 and 29.97 frames per second is handled by
dropping the first two frame indices (values 00 and 01) of every
minute, except every tenth minute. If the frame and the subframe
values are zero, they may be omitted. Subframes are measured in
hundredths of a frame.Examples:Normal Play Time (NPT) indicates the stream-absolute position
relative to the beginning of the presentation. The timestamp
consists of two parts: The mandatory first part may be expressed in
either seconds only or in hours, minutes, and seconds. The optional
second part consists of a decimal point and decimal figures and
indicates fractions of a second.The beginning of a presentation corresponds to 0.0 seconds.
Negative values are not defined.The special constant "now" is defined as the current instant of a
live event. It MAY only be used for live events and MUST NOT be used
for on-demand (i.e., non-live) content.NPT is defined as in Digital Storage Media Command and Control
(DSMb;CC) :
Intuitively, NPT is the clock the viewer associates with a
program. It is often digitally displayed on a DVD player. NPT
advances normally when in normal play mode (scale = 1), advances
at a faster rate when in fast-scan forward (high positive scale
ratio), decrements when in scan reverse (negative scale ratio)
and is fixed in pause mode. NPT is (logically) equivalent to
SMPTE time codes.Examples:The syntax is based on ISO
8601 and expresses the time elapsed since presentation start,
with two different notations allowed:The npt-hhmmss notation uses an ISO 8601 extended complete
representation of the time of the day format (Section 5.3.1.1 of
) using colons (":") as
separators between hours, minutes, and seconds (hh:mm:ss). The
hour counter is not limited to 0-24 hours; up to nineteen (19)
hour digits are allowed.In accordance with the requirements of the ISO 8601 time
format, the hours, minutes, and seconds MUST all be present,
with two digits used for minutes and for seconds and with at
least two digits for hours. An NPT of 7 minutes and 0
seconds is represented as "00:07:00", and an NPT of 392
hours, 0 minutes, and 6 seconds is represented as
"392:00:06".RTSP 1.0 allowed NPT in the npt-hhmmss notation without
any leading zeros to ensure that implementations don't fail;
for backward compatibility, all RTSP 2.0 implementations are
are REQUIRED to support receiving NPT values, hours,
minutes, or seconds, without leading zeros.The npt-sec notation expresses the time in seconds, using
between one and nineteen (19) digits.Both notations allow decimal fractions of seconds as
specified in Section 5.3.1.3 of ,
using at most nine digits, and allowing only "." (full stop) as the
decimal separator.The npt-sec notation is optimized for automatic generation; the
npt-hhmmss notation is optimized for consumption by human readers.
The "now" constant allows clients to request to receive the live
feed rather than the stored or time-delayed version. This is needed
since neither absolute time nor zero time are appropriate for this
case.Absolute time is expressed using a timestamp based on ISO 8601
. The date is a complete
representation of the calendar date in basic format (YYYYMMDD)
without separators (per Section 5.2.1.1 of ). The time of day is provided in the
complete representation basic format (hhmmss) as specified in
Section 5.3.1.1 of , allowing decimal
fractions of seconds following Section 5.3.1.3 requiring "." (full
stop) as decimal separator and limiting the number of digits to no
more than nine. The time expressed MUST use UTC (GMT), i.e., no time
zone offsets are allowed. The full date and time specification is
the eight-digit date followed by a "T" followed by the six-digit
time value, optionally followed by a full stop followed by one to
nine fractions of a second and ended by "Z", e.g.,
YYYYMMDDThhmmss.ssZ.The reasons for this time format rather than using "Date and Time on the Internet:
Timestamps" are historic. We continue to use the format
specified in RTSP 1.0. The motivations raised in RFC 3339 apply
to why a selection from ISO 8601 was made; however, a different
and even more restrictive selection was applied in this
case.Below are three examples of media time formats, first, a
request for a clock format range request for a starting time of
November 8, 1996 at 14 h 37 min and 20 1/4 seconds UTC playing for
10 min and 5 seconds, followed by a Media-Properties header's
"Time-Limited" UTC property for the 24th of December 2014 at 15
hours and 00 minutes, and finally a Terminate-Reason header "time"
property for the 18th of June 2013 at 16 hours, 12 minutes, and 56
seconds:Feature tags are unique identifiers used to designate features in
RTSP. These tags are used in Require (),
Proxy-Require (), Proxy-Supported
(), Supported (), and Unsupported () header fields.A feature tag definition MUST indicate which combination of
clients, servers, or proxies to which it applies.The creator of a new RTSP feature tag should either prefix the
feature tag with a reverse domain name (e.g.,
"com.example.mynewfeature" is an apt name for a feature whose inventor
can be reached at "example.com") or register the new feature tag with
the Internet Assigned Numbers Authority (IANA). (See , "IANA Considerations".)The usage of feature tags is further described in , which deals with capability handling.Message body tags are opaque strings that are used to compare two
message bodies from the same resource, for example, in caches or to
optimize setup after a redirect. Message body tags can be carried in
the MTag header (see ) or in SDP (see ). MTag is similar to ETag in HTTP/1.1 (see
Section 3.11 of ).A message body tag MUST be unique across all versions of all
message bodies associated with a particular resource. A given message
body tag value MAY be used for message bodies obtained by requests on
different URIs. The use of the same message body tag value in
conjunction with message bodies obtained by requests on different URIs
does not imply the equivalence of those message bodies.Message body tags are used in RTSP to make some methods
conditional. The methods are made conditional through the inclusion of
headers; see and for information on the If-Match and
If-None-Match headers, respectively. Note that RTSP message body tags
apply to the complete presentation, i.e., both the presentation
description and the individual media streams. Thus, message body tags
can be used to verify at setup time after a redirect that the same
session description applies to the media at the new location using the
If-Match header.When an RTSP server handles media, it is important to consider the
different properties a media instance for delivery and playback can
have. This specification considers the media properties listed below
in its protocol operations. They are derived from the differences
between a number of supported usages. Media that has a fixed (given) duration
that doesn't change during the lifetime of the RTSP session and is
known at the time of the creation of the session. It is expected
that the content of the media will not change, even if the
representation, such as encoding, or quality, may change.
Generally, one can seek, i.e., request any range, within the
media.This is a variation of the
on-demand case where external methods are used to manipulate the
actual content of the media setup for the RTSP session. The main
example is content defined by a playlist.Live media represents a progressing content
stream (such as broadcast TV) where the duration may or may not be
known. It is not seekable, only the content presently being
delivered can be accessed.A live stream that is combined
with a server-side capability to store and retain the content of
the live session and allow for random access delivery within the
part of the already-recorded content. The actual behavior of the
media stream is very much dependent on the retention policy for
the media stream; either the server will be able to capture the
complete media stream or it will have a limitation in how much
will be retained. The media range will dynamically change as the
session progress. For servers with a limited amount of storage
available for recording, there will typically be a sliding window
that moves forward while new data is made available and older data
is discarded.To cover the above usages, the following media properties with
appropriate values are specified.Random access is the ability to specify and get media delivered
starting from any time (instant) within the content, an operation
called "seeking". The Media-Properties header will indicate the
general capability for a media resource to perform random
access.The media is seekable to any out of
a large number of points within the media. Due to media-encoding
limitations, a particular point may not be reachable, but
seeking to a point close by is enabled. A floating-point number
of seconds may be provided to express the worst-case distance
between random access points.Seeking is only possible to the
beginning of the content.Seeking is not possible at all.If random access is possible, as indicated by the
Media-Properties header, the actual behavior policy when seeking can
be controlled using the Seek-Style
header.The following retention policies are used by media to limit
possible protocol operations:The media will not be removed as long
as the RTSP session is in existence.The media will not be removed before
the given wallclock time. After that time, it may or may not be
available anymore.The media (on fragment or unit
basis) will be retained for the specified duration.The media content and its timeline can be of different types,
e.g. pre-produced content on demand, a live source that is being
generated as time progresses, or something that is dynamically
altered or recomposed during playback. Therefore, a media property
for content modifications is needed and the following initial values
are defined:The content of the media will not
change, even if the representation, such as encoding or quality
changes.The content can change due to external
methods or triggers, such as playlists, but this will be
announced by explicit updates.As time progresses, new content
will become available. If the content is also retained, it will
become longer as everything between the start point and the
point currently being made available can be accessed. If the
media server uses a sliding-window policy for retention, the
start point will also change as time progresses.A particular media content item often supports only a limited set
or range of scales when delivering the media. To enable the client
to know what values or ranges of scale operations that the whole
content or the current position supports, a media properties
attribute for this is defined that contains a list with the values
or ranges that are supported. The attribute is named "Scales". The
"Scales" attribute may be updated at any point in the content due to
content consisting of spliced pieces or content being dynamically
updated by out-of-band mechanisms.This section shows examples of how one would map the above usages
to the properties and their values. Random Access:
Random-Access=5.0, Content Modifications: Immutable, Retention:
Unlimited or Time-Limited. Random
Access: Random-Access=3.0, Content Modifications: Dynamic,
Retention: Unlimited or Time-Limited. Random Access:
No-Seeking, Content Modifications: Time-Progressing, Retention:
Time-Duration=0.0 Random
Access: Random-Access=3.0, Content Modifications:
Time-Progressing, Retention: Time-Duration=7200.0RTSP is a text-based protocol that uses the ISO 10646 character set
in UTF-8 encoding per RFC 3629 . Lines MUST be
terminated by a CRLF.Text-based protocols make it easier to add optional parameters in
a self-describing manner. Since the number of parameters and the
frequency of commands is low, processing efficiency is not a
concern. Text-based protocols, if used carefully, also allow easy
implementation of research prototypes in scripting languages such as
Python, PHP, Perl and TCL.The ISO 10646 character set avoids character-set switching, but is
invisible to the application as long as US-ASCII is being used. This is
also the encoding used for text fields in RTCP.A request contains a method, the object the method is operating upon,
and parameters to further describe the method. Methods are idempotent
unless otherwise noted. Methods are also designed to require little or
no state maintenance at the media server.RTSP messages are either requests from client to server or from
server to client, and responses in the reverse direction. Request
() and response () messages use a format based on the generic
message format of RFC 5322 for transferring
bodies (the payload of the message). Both types of messages consist of
a start-line, zero or more header fields (also known as "headers"), an
empty line (i.e., a line with nothing preceding the CRLF) indicating
the end of the headers, and possibly the data of the message body. The
ABNF below is for illustration only; the
formal message specification is presented in .In the interest of robustness, agents MUST ignore any empty line(s)
received where a Request-Line or Status-Line is expected. In other
words, if the agent is reading the protocol stream at the beginning of
a message and receives any number of CRLFs first, it MUST ignore all
of the CRLFs.RTSP header fields (see ) include
general-header, request-header, response-header, and message body
header fields.The order in which header fields with differing field names are
received is not significant. However, it is "good practice" to send
general-header fields first, followed by a request-header or
response-header field, and ending with the message body header
fields.Multiple header fields with the same field-name MAY be present in a
message if and only if the entire field-value for that header field is
defined as a comma-separated list. It MUST be possible to combine the
multiple header fields into one "field-name: field-value" pair,
without changing the semantics of the message, by appending each
subsequent field-value to the first, each separated by a comma. The
order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field
value; thus, a proxy MUST NOT change the order of these field-values
when a message is forwarded.Unknown message headers MUST be ignored (skipping over the header
to the next protocol element, and not causing an error) by an RTSP
server or client. An RTSP proxy MUST forward unknown message headers.
Message headers defined outside of this specification that are
required to be interpreted by the RTSP agent will need to use feature tags and include them in the
appropriate Require or Proxy-Require header.The message body (if any) of an RTSP message is used to carry
further information for a particular resource associated with the
request or response. An example of a message body is an SDP
message.The presence of a message body in either a request or a response
MUST be signaled by the inclusion of a Content-Length header (see
) and Content-Type header (see
). A message body MUST NOT be
included in a request or response if the specification of the
particular method (see Method Definitions
) does not allow sending a message body. In case a message body
is received in a message when not expected, the message body data
SHOULD be discarded. This is to allow future extensions to define
optional use of a message body.An RTSP message that does not contain any message body is
terminated by the first empty line after the header fields (note: an
empty line is a line with nothing preceding the CRLF.). In RTSP
messages that contain message bodies, the empty line is followed by
the message body. The length of that body is determined by the value
of the Content-Length header.
The value in the header represents the length of the message body in
octets. If this header field is not present, a value of zero is
assumed, i.e., no message body present in the message. Unlike an HTTP
message, an RTSP message MUST contain a Content-Length header whenever
it contains a message body. Note that RTSP does not support the
HTTP/1.1 "chunked" transfer coding (see Section 4.1 of ).Given the moderate length of presentation descriptions
returned, the server should always be able to determine its
length, even if it is generated dynamically, making the chunked
transfer encoding unnecessary.General headers are headers that may be used in both requests and
responses. The general-headers are listed in :Header NameDefined inAccept-RangesCache-ControlConnectionCSeqDateMedia-PropertiesMedia-RangePipelined-RequestsProxy-SupportedRangeRTP-InfoScaleSeek-StyleServerSessionSpeedSupportedTimestampTransportUser-AgentViaA request message uses the format outlined below regardless of the
direction of a request, whether client to server or server to client:
Request line, containing the method to be applied to the
resource, the identifier of the resource, and the protocol version
in use;Zero or more Header lines, which can be of the following types:
general-headers (),
request-headers (), or message
body headers ();One empty line (CRLF) to indicate the end of the header
section;Optionally, a message body, consisting of one or more lines. The
length of the message body in octets is indicated by the
Content-Length message header.The request line provides the key information about the request:
what method, on what resources, and using which RTSP version. The
methods that are defined by this specification are listed in .MethodDefined inDESCRIBEGET_PARAMETEROPTIONSPAUSEPLAYPLAY_NOTIFYREDIRECTSETUPSET_PARAMETERTEARDOWNThe syntax of the RTSP request line has the following: <Method> SP <Request-URI> SP <RTSP-Version>
CRLF Note: This syntax cannot be freely changed in future
versions of RTSP. This line needs to remain parsable by older RTSP
implementations since it indicates the RTSP version of the
message.In contrast to HTTP/1.1 , RTSP requests
identify the resource through an absolute RTSP URI (including scheme,
host, and port) (see ) rather than just the
absolute path.HTTP/1.1 requires servers to understand the absolute URI, but
clients are supposed to use the Host request-header. This is
purely needed for backward compatibility with HTTP/1.0 servers, a
consideration that does not apply to RTSP.An asterisk "*" can be used instead of an absolute URI in the
Request-URI part to indicate that the request does not apply to a
particular resource but to the server or proxy itself, and is only
allowed when the request method does not necessarily apply to a
resource.For example: OPTIONS * RTSP/2.0An OPTIONS in this form will determine the capabilities of the
server or the proxy that first receives the request. If the capability
of the specific server needs to be determined, without regard to the
capability of an intervening proxy, the server should be addressed
explicitly with an absolute URI that contains the server's
address.For example: OPTIONS rtsp://example.com RTSP/2.0The RTSP headers in can be
included in a request, as request-headers, to modify the specifics of
the request.HeaderDefined inAcceptAccept-CredentialsAccept-EncodingAccept-LanguageAuthorizationBandwidthBlocksizeFromIf-MatchIf-Modified-SinceIf-None-MatchNotify-ReasonProxy-AuthorizationProxy-RequireReferrerRequest-StatusRequireTerminate-ReasonDetailed header definitions are provided in .New request-headers may be defined. If the receiver of the request
is required to understand the request-header, the request MUST include
a corresponding feature tag in a Require or Proxy-Require header to
ensure the processing of the header.After receiving and interpreting a request message, the recipient
responds with an RTSP response message. Normally, there is only one,
final, response. Responses using the response code class 1xx is the only
class for which there MAY be sent one or more responses prior to the
final response message.The valid response codes and the methods they can be used with are
listed in .The first line of a response message is the Status-Line, consisting
of the protocol version followed by a numeric status code and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the final
CRLF sequence.<RTSP-Version> SP <Status-Code> SP <Reason
Phrase> CRLFThe Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in . The reason phrase is
intended to give a short textual description of the Status-Code. The
Status-Code is intended for use by automata and the reason phrase is
intended for the human user. The client is not required to examine
or display the reason phrase.The first digit of the Status-Code defines the class of response.
The last two digits do not have any categorization role. There are
five values for the first digit: Informational - Request received, continuing
processSuccess - The action was successfully
received, understood, and acceptedRedirection - Further action needs to be
taken in order to complete the request (3rr rather than 3xx is
used as 304 is excluded; see )Client Error - The request contains bad
syntax or cannot be fulfilledServer Error - The server failed to fulfill
an apparently valid request The individual values of the numeric status codes defined
for RTSP 2.0, and an example set of corresponding reason phrases,
are presented in . The reason phrases
listed here are only recommended; they may be replaced by local
equivalents without affecting the protocol. Note that RTSP adopted
most HTTP/1.1 status codes and then added
RTSP-specific status codes starting at x50 to avoid conflicts with
future HTTP status codes that are desirable to import into RTSP. All
these codes are RTSP specific and RTSP has its own registry separate
from HTTP for status codes.RTSP status codes are extensible. RTSP applications are not
required to understand the meaning of all registered status codes,
though such understanding is obviously desirable. However,
applications MUST understand the class of any status code, as
indicated by the first digit, and treat any unrecognized response as
being equivalent to the x00 status code of that class, with an
exception for unknown 3xx codes, which MUST be treated as a 302
(Found). The reason for that exception is that the status code 300
(Multiple Choices in HTTP) is not defined for RTSP. A response with
an unrecognized status code MUST NOT be cached. For example, if an
unrecognized status code of 431 is received by the client, it can
safely assume that there was something wrong with its request and
treat the response as if it had received a 400 status code. In such
cases, user agents SHOULD present to the user the message body
returned with the response, since that message body is likely to
include human-readable information that will explain the unusual
status.CodeReasonMethod100Continueall200OKall301Moved Permanentlyall302Foundall303See Othern/a304Not Modifiedall305Use Proxyall400Bad Requestall401Unauthorizedall402Payment Requiredall403Forbiddenall404Not Foundall405Method Not Allowedall406Not Acceptableall407Proxy Authentication Requiredall408Request Timeoutall410Goneall412Precondition FailedDESCRIBE, SETUP413Request Message Body Too Largeall414Request-URI Too Longall415Unsupported Media Typeall451Parameter Not UnderstoodSET_PARAMETER, GET_PARAMETER452reservedn/a453Not Enough BandwidthSETUP454Session Not Foundall455Method Not Valid in This Stateall456Header Field Not Valid for Resourceall457Invalid RangePLAY, PAUSE458Parameter Is Read-OnlySET_PARAMETER459Aggregate Operation Not Allowedall460Only Aggregate Operation Allowedall461Unsupported Transportall462Destination Unreachableall463Destination ProhibitedSETUP464Data Transport Not Ready YetPLAY465Notification Reason UnknownPLAY_NOTIFY466Key Management Errorall470Connection Authorization Requiredall471Connection Credentials Not Acceptedall472Failure to Establish Secure Connectionall500Internal Server Errorall501Not Implementedall502Bad Gatewayall503Service Unavailableall504Gateway Timeoutall505RTSP Version Not Supportedall551Option Not Supportedall553Proxy UnavailableallThe response-headers allow the request recipient to pass additional
information about the response that cannot be placed in the
Status-Line. This header gives information about the server and about
further access to the resource identified by the Request-URI. All
headers currently classified as response-headers are listed in .HeaderDefined inAuthentication-InfoConnection-CredentialsLocationMTagProxy-AuthenticatePublicRetry-AfterUnsupportedWWW-AuthenticateResponse-header names can be extended reliably only in combination
with a change in the protocol version. However, the usage of feature
tags in the request allows the responding party to learn the
capability of the receiver of the response. A new or experimental
header can be given the semantics of response-header if all parties in
the communication recognize them to be a response-header. Unrecognized
headers in responses MUST be ignored.Some request and response messages include a message body, if not
otherwise restricted by the request method or response status code. The
message body consists of the content data itself (see also ).The SET_PARAMETER and GET_PARAMETER requests and responses, and the
DESCRIBE response as defined by this specification, can have a message
body; the purpose of the message body is defined in each case. All 4xx
and 5xx responses MAY also have a message body to carry additional
response information. Generally, a message body MAY be attached to any
RTSP 2.0 request or response, but the content of the message body MAY be
ignored by the receiver. Extensions to this specification can specify
the purpose and content of message bodies, including requiring their
inclusion.In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the message
body.Message body header fields define meta-information about the
content data in the message body. The message body header fields are
listed in .HeaderDefined inAllowContent-BaseContent-EncodingContent-LanguageContent-LengthContent-LocationContent-TypeExpiresLast-ModifiedThe extension-header mechanism allows additional message body
header fields to be defined without changing the protocol, but these
fields cannot be assumed to be recognizable by the recipient.
Unrecognized header fields MUST be ignored by the recipient and
forwarded by proxies.An RTSP message with a message body MUST include the Content-Type
and Content-Length headers. When a message body is included with a
message, the data type of that content data is determined via the
Content-Type and Content-Encoding header fields.Content-Type specifies the media type of the underlying data. There
is no default media format and the actual format used in the body is
required to be explicitly stated in the Content-Type header. By being
explicit and always requiring the inclusion of the Content-Type header
with accurate information, one avoids the many pitfalls in a
heuristic-based interpretation of the body content. The user
experience of HTTP and email have suffered from relying on such
heuristics.Content-Encoding may be used to indicate any additional
content-codings applied to the data, usually for the purpose of data
compression, that are a property of the requested resource. The
default encoding is 'identity', i.e. no transformation of the message
body.The Content-Length of a message is the length of the content,
measured in octets.The content format of the message body is provided using the Content-Type header. To enable the
responder of a request to determine which media type it should use,
the requester may include the Accept
header in a request to identify supported media types or media
type ranges suitable to the response. In case the responder is not
supporting any of the specified formats, then the request response
will be a 406 (Not Acceptable) error code.The media types that may be used on requests with message bodies
need to be determined through the use of feature tags, specification
requirement, or trial and error. Trial and error works because when
the responder does not support the media type of the message body, it
will respond with a 415 (Unsupported Media Type).The formats supported and their negotiation is done individually on
a per method and direction (request or response body) direction.
Requirements on supporting particular media types for use as message
bodies in requests and response SHALL also be specified on a
per-method and per-direction basis.RTSP messages are transferred between RTSP agents and proxies using a
transport connection. This transport connection uses TCP or TCP/TLS.
This transport connection is referred to as the "connection" or "RTSP
connection" within this document.RTSP requests can be transmitted using the two different connection
scenarios listed below: persistent - a transport connection is used for several
request/response transactions;transient - a transport connection is used for each single
request/response transaction.RFC 2326 attempted to specify an optional mechanism for transmitting
RTSP messages in connectionless mode over a transport protocol such as
UDP. However, it was not specified in sufficient detail to allow for
interoperable implementations. In an attempt to reduce complexity and
scope, and due to lack of interest, RTSP 2.0 does not attempt to define
a mechanism for supporting RTSP over UDP or other connectionless
transport protocols. A side effect of this is that RTSP requests MUST
NOT be sent to multicast groups since no connection can be established
with a specific receiver in multicast environments.Certain RTSP headers, such as the CSeq header (), which may appear to be relevant only to
connectionless transport scenarios, are still retained and MUST be
implemented according to this specification. In the case of CSeq, it is
quite useful for matching responses to requests if the requests are
pipelined (see ). It is also useful in
proxies for keeping track of the different requests when aggregating
several client requests on a single TCP connection.Since RTSP messages are transmitted using reliable transport
protocols, they MUST NOT be retransmitted at the RTSP level. Instead,
the implementation must rely on the underlying transport to provide
reliability. The RTSP implementation may use any indication of
reception acknowledgment of the message from the underlying transport
protocols to optimize the RTSP behavior.If both the underlying reliable transport, such as TCP, and the
RTSP application retransmit requests, each packet loss or message
loss may result in two retransmissions. The receiver typically
cannot take advantage of the application-layer retransmission
since the transport stack will not deliver the application-layer
retransmission before the first attempt has reached the receiver.
If the packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the
congestion.Lack of acknowledgment of an RTSP request should be handled within
the constraints of the connection timeout considerations described
below ().A TCP transport can be used for both persistent connections (for
several message exchanges) and transient connections (for a single
message exchange). Implementations of this specification MUST support
RTSP over TCP. The scheme of the RTSP URI ()
allows the client to specify the port it will contact the server on,
and defines the default port to use if one is not explicitly
given.In addition to the registered default ports, i.e., 554 (rtsp) and
322 (rtsps), there is an alternative port 8554 registered. This port
may provide some benefits over non-registered ports if an RTSP server
is unable to use the default ports. The benefits may include
preconfigured security policies as well as classifiers in network
monitoring tools.An RTSP client opening a TCP connection to access a particular
resource as identified by a URI uses the IP address and port derived
from the host and port parts of the URI. The IP address is either the
explicit address provided in the URI or any of the addresses provided
when performing A and AAAA record DNS lookups of the hostname in the
URI.A server MUST handle both persistent and transient connections.Transient connections facilitate mechanisms for fault
tolerance. They also allow for application-layer mobility. A
server-and-client pair that supports transient connections can
survive the loss of a TCP connection; e.g., due to a NAT timeout.
When the client has discovered that the TCP connection has been
lost, it can set up a new one when there is need to communicate
again.A persistent connection is RECOMMENDED to be used for all
transactions between the server and client, including messages for
multiple RTSP sessions. However, a persistent connection MAY be closed
after a few message exchanges. For example, a client may use a
persistent connection for the initial SETUP and PLAY message exchanges
in a session and then close the connection. Later, when the client
wishes to send a new request, such as a PAUSE for the session, a new
connection would be opened. This connection may be either transient or
persistent.An RTSP agent MAY use one connection to handle multiple RTSP
sessions on the same server. The RTSP agent SHALL NOT use more than
one connection per RTSP session at any given point.Having only one connection in use at any time avoids confusion
regarding on which connection any server-to-client requests shall
be sent. Using a single connection for multiple RTSP sessions also
saves complexity by enabling the server to maintain less state
about its connection resources on the server. Not using more than
one connection at a time for a particular RTSP session avoids
wasting connection resources and allows the server to track only
the most recently used client-to-server connection for each RTSP
session as being the currently valid server-to-client
connection.RTSP allows a server to send requests to a client. However, this
can be supported only if a client establishes a persistent connection
with the server. In cases where a persistent connection does not exist
between a server and its client, due to the lack of a signaling
channel, the server may be forced to silently discard RTSP messages,
and it may even drop an RTSP session without notifying the client. An
example of such a case is when the server desires to send a REDIRECT
request for an RTSP session to the client but is not able to do so
because it cannot reach the client. A server that attempts to send a
request to a client that has no connection currently to the server
SHALL discard the request. Without a persistent connection between the client and the
server, the media server has no reliable way of reaching the
client. Because of the likely failure of server-to-client
established connections, the server will not even attempt
establishing any connection.Queuing of server-to-client requests has been considered.
However, a security issue exists as to how it might be possible to
authorize a client establishing a new connection as being a
legitimate receiver of a request related to a particular RTSP
session, without the client first issuing requests related to the
pending request. Thus, it would be likely to make any such
requests even more delayed and less useful.The sending of client and server requests can be asynchronous
events. To avoid deadlock situations, both client and server MUST be
able to send and receive requests simultaneously. As an RTSP response
may be queued up for transmission, reception or processing behind the
peer RTSP agent's own requests, all RTSP agents are required to have a
certain capability of handling outstanding messages. A potential issue
is that outstanding requests may time out despite being processed by
the peer; this can be due to the response being caught in the queue
behind a number of requests that the RTSP agent is processing but that
take some time to complete. To avoid this problem, an RTSP agent
should buffer incoming messages locally so that any response messages
can be processed immediately upon reception. If responses are
separated from requests and directly forwarded for processing, not
only can the result be used immediately, the state associated with
that outstanding request can also be released. However, buffering a
number of requests on the receiving RTSP agent consumes resources and
enables a resource exhaustion attack on the agent. Therefore, this
buffer should be limited so that an unreasonable number of requests or
total message size is not allowed to consume the receiving agent's
resources. In most APIs, having the receiving agent stop reading from
the TCP socket will result in TCP's window being clamped, thus forcing
the buffering onto the sending agent when the load is larger than
expected. However, as both RTSP message sizes and frequency may be
changed in the future by protocol extensions, an agent should be
careful about taking harsher measurements against a potential attack.
When under attack, an RTSP agent can close TCP connections and release
state associated with that TCP connection.To provide some guidance on what is reasonable, the following
guidelines are given. It is RECOMMENDED that: an RTSP agent should not have more than 10 outstanding requests
per RTSP session;an RTSP agent should not have more than 10 outstanding requests
that are not related to an RTSP session or that are requesting to
create an RTSP session.In light of the above, it is RECOMMENDED that clients use
persistent connections whenever possible. A client that supports
persistent connections MAY "pipeline" its requests (see ).RTSP agents can send requests to multiple different destinations,
either server or client contexts over the same connection to a proxy.
Then, the proxy forks the message to the different destinations over
proxy-to-agent connections. In these cases when multiple requests are
outstanding, the requesting agent MUST be ready to receive the
responses out of order compared to the order they where sent on the
connection. The order between multiple messages for each destination
will be maintained; however, the order between response from different
destinations can be different.The reason for this is to avoid a head-of-line blocking
situation. In a sequence of requests, an early outstanding request
may take time to be processed at one destination. Simultaneously,
a response from any other destination that was later in the
sequence of requests may have arrived at the proxy; thus, allowing
out-of-order responses avoids forcing the proxy to buffer this
response and instead deliver it as soon as possible. Note, this
will not affect the order in which the messages sent to each
separate destination were processed at the request
destination.This scenario can occur in two cases involving proxies. The first
is a client issuing requests for sessions on different servers using a
common client-to-proxy connection. The second is for server-to-client
requests, like REDIRECT being sent by the server over a common
transport connection the proxy created for its different connecting
clients.The client MAY close a connection at any point when no outstanding
request/response transactions exist for any RTSP session being managed
through the connection. The server, however, SHOULD NOT close a
connection until all RTSP sessions being managed through the
connection have been timed out (). A
server SHOULD NOT close a connection immediately after responding to a
session-level TEARDOWN request for the last RTSP session being
controlled through the connection. Instead, the server should wait for
a reasonable amount of time for the client to receive and act upon the
TEARDOWN response and then initiate the connection closing. The server
SHOULD wait at least 10 seconds after sending the TEARDOWN response
before closing the connection.This is to ensure that the client has time to issue a SETUP for
a new session on the existing connection after having torn the
last one down. Ten seconds should give the client ample
opportunity to get its message to the server.A server SHOULD NOT close the connection directly as a result of
responding to a request with an error code.Certain error responses such as 460 (Only Aggregate Operation
Allowed) () are used for negotiating
capabilities of a server with respect to content or other factors.
In such cases, it is inefficient for the server to close a
connection on an error response. Also, such behavior would prevent
implementation of advanced or special types of requests or result
in extra overhead for the client when testing for new features. On
the other hand, keeping connections open after sending an error
response poses a Denial-of-Service (DoS) security risk ().The server MAY close a connection if it receives an incomplete
message and if the message is not completed within a reasonable amount
of time. It is RECOMMENDED that the server wait at least 10 seconds
for the completion of a message or for the next part of the message to
arrive (which is an indication that the transport and the client are
still alive). Servers believing they are under attack or that are
otherwise starved for resources during that event MAY consider using a
shorter timeout.If a server closes a connection while the client is attempting to
send a new request, the client will have to close its current
connection, establish a new connection, and send its request over the
new connection.An RTSP message SHOULD NOT be terminated by closing the connection.
Such a message MAY be considered to be incomplete by the receiver and
discarded. An RTSP message is properly terminated as defined in .Receivers of a request (responders) SHOULD respond to requests in a
timely manner even when a reliable transport such as TCP is used.
Similarly, the sender of a request (requester) SHOULD wait for a
sufficient time for a response before concluding that the responder
will not be acting upon its request.A responder SHOULD respond to all requests within 5 seconds. If the
responder recognizes that the processing of a request will take longer
than 5 seconds, it SHOULD send a 100 (Continue) response as soon as
possible. It SHOULD continue sending a 100 response every 5 seconds
thereafter until it is ready to send the final response to the
requester. After sending a 100 response, the responder MUST send a
final response indicating the success or failure of the request.A requester SHOULD wait at least 10 seconds for a response before
concluding that the responder will not be responding to its request.
After receiving a 100 response, the requester SHOULD continue waiting
for further responses. If more than 10 seconds elapse without
receiving any response, the requester MAY assume that the responder is
unresponsive and abort the connection by closing the TCP
connection.In some cases, multiple RTSP sessions share the same transport
connection; abandoning a request and closing the connection may have
significant impact on those other sessions. First of all, other RTSP
requests may have become queued up due to the request taking a long
time to process. Secondly, those sessions also lose the possibility to
receive server-to-client requests. To mitigate that situation, the
RTSP client or server SHOULD establish a new connection and send any
requests that are queued up or that haven't received a response on
this new connection. Thirdly, to ensure that the RTSP server knows
which connection is valid for a particular RTSP session, the RTSP
agent SHOULD send a keep-alive request, if no other request will be
sent immediately for that RTSP session, for each RTSP session on the
old connection. The keep-alive request will normally be a
SET_PARAMETER with a session header to inform the server that this
agent cares about this RTSP session.A requester SHOULD wait longer than 10 seconds for a response if it
is experiencing significant transport delays on its connection to the
responder. The requester is capable of determining the Round-Trip Time
(RTT) of the request/response cycle using the Timestamp header () in any RTSP request.The 10-second wait was chosen for the following reasons. It
gives TCP time to perform a couple of retransmissions, even if
operating on default values. It is short enough that users may not
abandon the process themselves. However, it should be noted that
10 seconds can be aggressive on certain types of networks. The
5-second value for 1xx messages is half the timeout giving a
reasonable chance of successful delivery before timeout happens on
the requester side.RTSP requires the client to periodically show its liveness to the
server or the server may terminate any session state. Several
different protocol mechanism include in their usage a liveness proof
from the client. These mechanisms are RTSP requests with a Session
header to the server; if RTP & RTCP is used for media data
transport and the transport is established, the RTCP message proves
liveness; or through any other used media-transport protocol capable
of indicating liveness of the RTSP client. It is RECOMMENDED that a
client not wait to the last second of the timeout before trying to
send a liveness message. The RTSP message may take some time to arrive
safely at the receiver, due to packet loss and TCP retransmissions. To
show liveness between RTSP requests being issued to accomplish other
things, the following mechanisms can be used, in descending order of
preference: If RTP is used for media transport, RTCP
SHOULD be used. If RTCP is used to report transport statistics, it
will necessarily also function as a keep-alive. The server can
determine the client by network address and port together with the
fact that the client is reporting on the server's RTP sender
sources (synchronization source (SSRCs)). A downside of using RTCP
is that it only gives statistical guarantees of reaching the
server. However, the probability of a false client timeout is so
low that it can be ignored in most cases. For example, assume a
session with a 60-second timeout and enough bitrate assigned to
RTCP messages to send a message from client to server on average
every 5 seconds. That client has, for a network with 5% packet
loss, a probability of failing to confirm liveness within the
timeout interval for that session of 2.4*E-16. Sessions with
shorter timeouts, much higher packet loss, or small RTCP
bandwidths SHOULD also implement one or more of the mechanisms
below.When using SET_PARAMETER for
keep-alives, a body SHOULD NOT be included. This method is the
RECOMMENDED RTSP method to use for a request intended only to
perform keep-alives. RTSP servers MUST support the SET_PARAMETER
method, so that clients can always use this mechanism.When using GET_PARAMETER for
keep-alives, a body SHOULD NOT be included, dependent on
implementation support in the server. Use the OPTIONS method to
determine if there is method support or simply try.This method is also usable, but it causes
the server to perform more unnecessary processing and results in
bigger responses than necessary for the task. The reason is that
the server needs to determine the capabilities associated with the
media resource to correctly populate the Public and Allow
headers.The timeout parameter of the Session
header MAY be included in a SETUP response and MUST NOT be
included in requests. The server uses it to indicate to the client how
long the server is prepared to wait between RTSP commands or other
signs of life before closing the session due to lack of activity (see
). The timeout is measured in seconds,
with a default of 60 seconds. The length of the session timeout MUST
NOT be changed in an established session.Explicit IPv6 support was not present
in RTSP 1.0. RTSP 2.0 has been updated for explicit IPv6 support.
Implementations of RTSP 2.0 MUST understand literal IPv6 addresses in
URIs and RTSP headers. Although the general URI format envisages
potential future new versions of the literal IP address, usage of any
such new version would require other modifications to the RTSP
specification (e.g., address fields in the Transport header).Overload in RTSP can occur when servers and proxies have
insufficient resources to complete the processing of a request. An
improper handling of such an overload situation at proxies and servers
can impact the operation of the RTSP deployment, and probably worsen
the situation. RTSP defines the 503 (Service Unavailable) response
() to let servers and proxies notify
requesting proxies and RTSP clients about an overload situation. In
conjunction with the Retry-After header (), the server or proxy can indicate the time
after which the requesting entity can send another request to the
proxy or server.There are two scopes of such 503 answers. The first scope is for an
established RTSP session, where the request resulting in the 503
response as well as the response itself carries a Session header
identifying the session that is suffering overload. This response only
applies to this particular session. The other scope is the general
RTSP server as identified by the host in the Request-URI. Such a 503
answer with any Retry-After header applies to all requests that are
not session specific to that server, including a SETUP request
intended to create a new RTSP session.Another scope for overload situations exists: the RTSP proxy. To
enable an RTSP proxy to signal that it is overloaded, or otherwise
unavailable and unable to handle the request, a 553 response code has
been defined with the meaning "Proxy Unavailable". As with servers,
there is a separation in response scopes between requests associated
with existing RTSP sessions and requests to create new sessions or
general proxy requests.Simply implementing and using the 503 (Service Unavailable) and 553
(Proxy Unavailable) response codes is not sufficient for properly
handling overload situations. For instance, a simplistic approach
would be to send the 503 response with a Retry-After header set to a
fixed value. However, this can cause a situation in which multiple
RTSP clients again send requests to a proxy or server at roughly the
same time, which may again cause an overload situation. Another
situation would be if the "old" overload situation is not yet
resolved, i.e., the length indicated in the Retry-After header was too
short for the overload situation to subside.An RTSP server or proxy in an overload situation must select the
value of the Retry-After header carefully, bearing in mind its current
load situation. It is REQUIRED to increase the timeout period in
proportion to the current load on the server, i.e., an increasing
workload should result in an increased length of the indicated
unavailability. It is REQUIRED not to send the same value in the
Retry-After header to all requesting proxies and clients, but to add a
variation to the mean value of the Retry-After header.A more complex case may arise when a load-balancing RTSP proxy is
in use. This is the case when an RTSP proxy is used to select amongst
a set of RTSP servers to handle the requests or when multiple server
addresses are available for a given server name. The proxy or client
may receive a 503 (Service Unavailable) or 553 (Proxy Unavailable)
response code from one of its RTSP servers or proxies, or a TCP
timeout (if the server is even unable to handle the request message).
The proxy or client simply retries the other addresses or configured
proxies, but it may also receive a 503 (Service Unavailable) or 553
(Proxy Unavailable) response or TCP timeouts from those addresses. In
such a situation, where none of the RTSP servers/proxies/addresses can
handle the request, the RTSP agent has to wait before it can send any
new requests to the RTSP server. Any additional request to a specific
address MUST be delayed according to the Retry-After headers received.
For addresses where no response was received or TCP timeout occurred,
an initial wait timer SHOULD be set to 5 seconds. That timer MUST be
doubled for each additional failure to connect or receive response
until the value exceeds 30 minutes when the timer's mean value may be
set to 30 minutes. It is REQUIRED not to set the same value in the
timer for each scheduling, but instead to add a variation to the mean
value, resulting in picking a random value within the range of 0.5 to
1.5 times the mean value.This section describes the available capability-handling mechanism
that allows RTSP to be extended. Extensions to this version of the
protocol are basically done in two ways. Firstly, new headers can be
added. Secondly, new methods can be added. The capability-handling
mechanism is designed to handle both cases.When a method is added, the involved parties can use the OPTIONS
method to discover whether it is supported. This is done by issuing an
OPTIONS request to the other party. Depending on the URI, it will either
apply in regard to a certain media resource, the whole server in
general, or simply the next hop. The OPTIONS response MUST contain a
Public header that declares all methods supported for the indicated
resource.It is not necessary to use OPTIONS to discover support of a method,
as the client could simply try the method. If the receiver of the
request does not support the method, it will respond with an error code
indicating the method is either not implemented (501) or does not apply
for the resource (405). The choice between the two discovery methods
depends on the requirements of the service.Feature tags are defined to handle functionality additions that are
not new methods. Each feature tag represents a certain block of
functionality. The amount of functionality that a feature tag represents
can vary significantly. For example, a feature tag can represent the
functionality a single RTSP header provides. Another feature tag can
represent much more functionality, such as the "play.basic" feature tag, which represents
the minimal media delivery for playback implementation.Feature tags are used to determine whether the client, server, or
proxy supports the functionality that is necessary to achieve the
desired service. To determine support of a feature tag, several
different headers can be used, each explained below: This header is used to determine the
complete set of functionality that both client and server have, in
general, and is not dependent on a specific resource. The intended
usage is to determine before one needs to use a functionality that
it is supported. It can be used in any method, but OPTIONS is the
most suitable as it simultaneously determines all methods that are
implemented. When sending a request, the requester declares all its
capabilities by including all supported feature tags. This results
in the receiver learning the requester's feature support. The
receiver then includes its set of features in the response.This header is used in a similar
fashion as the Supported header, but instead of giving the supported
functionality of the client or server, it provides both the
requester and the responder a view of the common functionality
supported in general by all members of the proxy chain between the
client and server; it does not depend on the resource. Proxies are
required to add this header whenever the Supported header is
present, but proxies may also add it independently of the
requester.This header can be included in any request
where the endpoint, i.e., the client or server, is required to
understand the feature to correctly perform the request. This can,
for example, be a SETUP request, where the server is required to
understand a certain parameter to be able to set up the media
delivery correctly. Ignoring this parameter would not have the
desired effect and is not acceptable. Therefore, the endpoint
receiving a request containing a Require MUST negatively acknowledge
any feature that it does not understand and not perform the request.
The response in cases where features are not supported is 551
(Option Not Supported). Also, the features that are not supported
are given in the Unsupported header in the response.This header has the same purpose and
behavior as Require except that it only applies to proxies and not
the endpoint. Features that need to be supported by both proxies and
endpoints need to be included in both the Require and Proxy-Require
header.This header is used in a 551 (Option Not
Supported) error response, to indicate which features were not
supported. Such a response is only the result of the usage of the
Require or Proxy-Require headers where one or more features were not
supported. This information allows the requester to make the best of
situations as it knows which features are not supported.An implementation supporting all normative parts of this
specification for the setup and control of playback of media uses the
feature tag "play.basic" to indicate this support. The appendices
(starting with letters) are not part of the functionality included in
the feature tag unless the appendix is explicitly specified in a main
section as being a required appendix.Note: This feature tag does not mandate any media delivery
protocol, such as RTP.In RTSP 1.0, there was a minimal implementation section.
However, that was not consistent with the rest of the
specification. So, rather than making an attempt to explicitly
enumerate the features for play.basic, this specification has to
be taken as a whole and the necessary features normatively defined
as being required are included.Pipelining is a general method to improve performance of
request/response protocols by allowing the requesting agent to have more
than one request outstanding and to send them over the same persistent
connection. For RTSP, where the relative order of requests will matter,
it is important to maintain the order of the requests. Because of this,
the responding agent MUST process the incoming requests in their sending
order. The sending order can be determined by the CSeq header and its
sequence number. For TCP, the delivery order will be the same, between
two agents, as the sending order. The processing of the request MUST
also have been finished before processing the next request from the same
agent. The responses MUST be sent in the order the requests were
processed.RTSP 2.0 has extended support for pipelining beyond the capabilities
in RTSP 1.0. As a major improvement, all requests involved in setting up
and initiating media delivery can now be pipelined, indicated by the
Pipelined-Request header (see ).
This header allows a client to request that two or more requests be
processed in the same RTSP session context that the first request
creates. In other words, a client can request that two or more media
streams be set up and then played without needing to wait for a single
response. This speeds up the initial start-up time for an RTSP session
by at least one RTT.If a pipelined request builds on the successful completion of one or
more prior requests, the requester must verify that all requests were
executed as expected. A common example will be two SETUP requests and a
PLAY request. In case one of the SETUP requests fails unexpectedly, the
PLAY request can still be successfully executed. However, the resulting
presentation will not be as expected by the requesting client, as only a
single media instead of two will be played. In this case, the client can
send a PAUSE request, correct the failing SETUP request, and then
request it be played.The method indicates what is to be performed on the resource
identified by the Request-URI. The method name is case sensitive. New
methods may be defined in the future. Method names MUST NOT start with a
$ character (decimal 36) and MUST be a token as defined by the ABNF
in . The methods are
summarized in .methoddirectionobjectServer req.Client req.DESCRIBEC -> SP,SrecommendedrecommendedGET_PARAMETERC -> SP,SoptionaloptionalS -> CP,SoptionaloptionalOPTIONSC -> SP,SrequiredrequiredS -> CP,SoptionaloptionalPAUSEC -> SP,SrequiredrequiredPLAYC -> SP,SrequiredrequiredPLAY_NOTIFYS -> CP,SrequiredrequiredREDIRECTS -> CP,SoptionalrequiredSETUPC -> SSrequiredrequiredSET_PARAMETERC -> SP,SrequiredoptionalS -> CP,SoptionaloptionalTEARDOWNC -> SP,SrequiredrequiredS -> CPrequiredrequiredNote on : This table covers RTSP
methods, their direction, and on what objects (P: presentation, S:
stream) they operate. Further, it indicates whether a server or a
client implementation is required (mandatory), recommended, or
optional.Further note on : the GET_PARAMETER
is optional. For example, a fully functional server can be built to
deliver media without any parameters. However, SET_PARAMETER is
required, i.e., mandatory to implement for the server; this is due
to its usage for keep-alive. PAUSE is required because it is the
only way of leaving the Play state without terminating the whole
session.If an RTSP agent does not support a particular method, it MUST return
a 501 (Not Implemented) response code and the requesting RTSP agent, in
turn, SHOULD NOT try this method again for the given agent/resource
combination. An RTSP proxy whose main function is to log or audit and
not modify transport or media handling in any way MAY forward RTSP
messages with unknown methods. Note that the proxy still needs to
perform the minimal required processing, like adding the Via header.The semantics of the RTSP OPTIONS method is similar to that of the
HTTP OPTIONS method described in Section 4.3.7 of . However, in RTSP, OPTIONS is bidirectional in that
a client can send the request to a server and vice versa. A client
MUST implement the capability to send an OPTIONS request and a server
or a proxy MUST implement the capability to respond to an OPTIONS
request. In addition to this "MUST-implement" functionality, clients,
servers and proxies MAY provide support both for sending OPTIONS
requests and for generating responses to the requests.An OPTIONS request may be issued at any time. Such a request does
not modify the session state. However, it may prolong the session
lifespan (see below). The URI in an OPTIONS request determines the
scope of the request and the corresponding response. If the
Request-URI refers to a specific media resource on a given host, the
scope is limited to the set of methods supported for that media
resource by the indicated RTSP agent. A Request-URI with only the host
address limits the scope to the specified RTSP agent's general
capabilities without regard to any specific media. If the Request-URI
is an asterisk ("*"), the scope is limited to the general capabilities
of the next hop (i.e., the RTSP agent in direct communication with the
request sender).Regardless of the scope of the request, the Public header MUST
always be included in the OPTIONS response, listing the methods that
are supported by the responding RTSP agent. In addition, if the scope
of the request is limited to a media resource, the Allow header MUST
be included in the response to enumerate the set of methods that are
allowed for that resource unless the set of methods completely matches
the set in the Public header. If the given resource is not available,
the RTSP agent SHOULD return an appropriate response code, such as 3rr
or 4xx. The Supported header MAY be included in the request to query
the set of features that are supported by the responding RTSP
agent.The OPTIONS method can be used to keep an RTSP session alive.
However, this is not the preferred way of session keep-alive
signaling; see . An OPTIONS request
intended for keeping alive an RTSP session MUST include the Session
header with the associated session identifier. Such a request SHOULD
also use the media or the aggregated control URI as the
Request-URI.Example:Note that the "gzipped-messages" feature tag in the Proxy-Require
is a fictitious feature.The DESCRIBE method is used to retrieve the description of a
presentation or media object from a server. The Request-URI of the
DESCRIBE request identifies the media resource of interest. The client
MAY include the Accept header in the request to list the description
formats that it understands. The server MUST respond with a
description of the requested resource and return the description in
the message body of the response, if the DESCRIBE method request can
be successfully fulfilled. The DESCRIBE reply-response pair
constitutes the media initialization phase of RTSP.The DESCRIBE response SHOULD contain all media initialization
information for the resource(s) that it describes. Servers SHOULD NOT
use the DESCRIBE response as a means of media indirection by having
the description point at another server; instead, using the 3rr
responses is RECOMMENDED.By forcing a DESCRIBE response to contain all media
initialization information for the set of streams that it
describes, and discouraging the use of DESCRIBE for media
indirection, any looping problems can be avoided that might have
resulted from other approaches.Example:Media initialization is a requirement for any RTSP-based system,
but the RTSP specification does not dictate that this is required to
be done via the DESCRIBE method. There are three ways that an RTSP
client may receive initialization information: via an RTSP DESCRIBE requestvia some other protocol (HTTP, email attachment, etc.)via some form of user interfaceIf a client obtains a valid description from an alternate source,
the client MAY use this description for initialization purposes
without issuing a DESCRIBE request for the same media. The client
should use any MTag to either validate the presentation description or
make the session establishment conditional on being valid.It is RECOMMENDED that minimal servers support the DESCRIBE method,
and highly recommended that minimal clients support the ability to act
as "helper applications" that accept a media initialization file from
a user interface, or other means that are appropriate to the operating
environment of the clients.The description below uses the following states in a protocol state
machine that is related to a specific session when that session has
been created. The state transitions are driven by protocol
interactions. For additional information about the state machine, see
.Initial state. No session exists.Session is ready to start playing.Session is playing, i.e., sending media-stream
data in the direction S->C.The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. The SETUP method may be used in two
different cases, namely, creating an RTSP session and changing the
transport parameters of media streams that are already set up. SETUP
can be used in all three states, Init, Ready, and Play, to change the
transport parameters. Additionally, Init and Ready can also be used
for the creation of the RTSP session. The usage of the SETUP method in
the Play state to add a media resource to the session is
unspecified.The Transport header, see , specifies
the media-transport parameters acceptable to the client for data
transmission; the response will contain the transport parameters
selected by the server. This allows the client to enumerate, in
descending order of preference, the transport mechanisms and
parameters acceptable to it, so the server can select the most
appropriate. It is expected that the session description format used
will enable the client to select a limited number of possible
configurations that are offered as choices to the server. All
transport-related parameters SHALL be included in the Transport
header; the use of other headers for this purpose is NOT RECOMMENDED
due to middleboxes, such as firewalls or NATs.For the benefit of any intervening firewalls, a client MUST
indicate the known transport parameters, even if it has no influence
over these parameters, for example, where the server advertises a
fixed-multicast address as destination.Since SETUP includes all transport initialization information,
firewalls and other intermediate network devices (which need this
information) are spared the more arduous task of parsing the
DESCRIBE response, which has been reserved for media
initialization.The client MUST include the Accept-Ranges header in the request,
indicating all supported unit formats in the Range header. This allows
the server to know which formats it may use in future session-related
responses, such as a PLAY response without any range in the request.
If the client does not support a time format necessary for the
presentation, the server MUST respond using 456 (Header Field Not
Valid for Resource) and include the Accept-Ranges header with the
range unit formats supported for the resource.In a SETUP response, the server MUST include the Accept-Ranges
header (see ) to indicate which time
formats are acceptable to use for this media resource.The SETUP 200 OK response MUST include the Media-Properties header
(see ). The combination of the
parameters of the Media-Properties header indicates the nature of the
content present in the session (see also ). For example, a live stream
with time shifting is indicated byRandom access set to Random-Access,Content Modifications set to Time-Progressing, andRetention set to Time-Duration (with specific recording window
time value).The SETUP 200 OK response MUST include the Media-Range header (see
) if the media is
Time-Progressing.A basic example for SETUP:In the above example, the client wants to create an RTSP session
containing the media resource "rtsp://example.com/foo/bar/baz.rm". The
transport parameters acceptable to the client are either RTP/AVP/UDP
(UDP per default) to be received on client port 4588 and 4589 at the
address the RTSP setup connection comes from or RTP/AVP interleaved on
the RTSP control channel. The server selects the RTP/AVP/UDP transport
and adds the address and ports it will send and receive RTP and RTCP
from, and the RTP SSRC that will be used by the server.The server MUST generate a session identifier in response to a
successful SETUP request unless a SETUP request to a server includes a
session identifier or a Pipelined-Requests header referencing an
existing session context. In that latter case, the server MUST bundle
this SETUP request into the existing session (aggregated session) or
return a 459 (Aggregate Operation Not Allowed) error code (see ). An aggregate control URI MUST be used to
control an aggregated session. This URI MUST be different from the
stream control URIs of the individual media streams included in the
aggregate (see for aggregated
sessions and for the particular URIs see ). The aggregate control URI is to be
specified by the session description if the server supports aggregated
control and aggregated control is desired for the session. However,
even if aggregated control is offered, the client MAY choose not to
set up the session in aggregated control. If an aggregate control URI
is not specified in the session description, it is normally an
indication that non-aggregated control should be used. The SETUP of
media streams in an aggregate that has not been given an aggregated
control URI is unspecified.While the session ID sometimes carries enough information for
aggregate control of a session, the aggregate control URI is still
important for some methods such as SET_PARAMETER where the control
URI enables the resource in question to be easily identified. The
aggregate control URI is also useful for proxies, enabling them to
route the request to the appropriate server, and for logging,
where it is useful to note the actual resource on which a request
was operating.A session will exist until it is either removed by a TEARDOWN
request or is timed out by the server. The server MAY remove a session
that has not demonstrated liveness signs from the client(s) within a
certain timeout period. The default timeout value is 60 seconds; the
server MAY set this to a different value and indicate so in the
timeout field of the Session header in the SETUP response. For further
discussion, see . Signs of liveness for an
RTSP session include any RTSP requests from a client that contain a
Session header with the ID for that session, as well as RTCP sender or
receiver reports if RTP is used to transport the underlying media
stream. RTCP sender reports may, for example, be received in session
where the server is invited into a conference session and are thus
valid as a liveness indicator.If a SETUP request on a session fails for any reason, the session
state, as well as transport and other parameters for associated
streams, MUST remain unchanged from their values as if the SETUP
request had never been received by the server.A client MAY issue a SETUP request for a stream that is already
set up or playing in the session to change transport parameters,
which a server MAY allow. If it does not allow the changing of
parameters, it MUST respond with error 455 (Method Not Valid in This
State). The reasons to support changing transport parameters include
allowing application-layer mobility and flexibility to utilize the
best available transport as it becomes available. If a client
receives a 455 error when trying to change transport parameters
while the server is in Play state, it MAY try to put the server in
Ready state using PAUSE before issuing the SETUP request again. If
that also fails, the changing of transport parameters will require
that the client perform a TEARDOWN of the affected media and then
set it up again. For an aggregated session, not tearing down all the
media at the same time will avoid the creation of a new session.All transport parameters MAY be changed. However, the primary
usage expected is to either change the transport protocol
completely, like switching from Interleaved TCP mode to UDP or vice
versa, or to change the delivery address.In a SETUP response for a request to change the transport
parameters while in Play state, the server MUST include the Range
header to indicate at what point the new transport parameters will
be used. Further, if RTP is used for delivery, the server MUST also
include the RTP-Info header to indicate at what timestamp and RTP
sequence number the change will take place. If both RTP-Info and
Range are included in the response, the "rtp_time" parameter and
start point in the Range header MUST be for the corresponding time,
i.e., be used in the same way as for PLAY to ensure the correct
synchronization information is available.If the transport-parameters change that happened while in Play
state results in a change of synchronization-related information,
for example, changing RTP SSRC, the server MUST include the
necessary synchronization information in the SETUP response.
However, the server SHOULD avoid changing the synchronization
information if possible.This section describes the usage of the PLAY method in general, for
aggregated sessions, and in different usage scenarios.The PLAY method tells the server to start sending data via the
mechanism specified in SETUP and which part of the media should be
played out. PLAY requests are valid when the session is in Ready or
Play state. A PLAY request MUST include a Session header to indicate
to which session the request applies.Upon receipt of the PLAY request, the server MUST position the
normal play time to the beginning of the range specified in the
received Range header, within the limits of the media resource and
in accordance with the Seek-Style
header. It MUST deliver stream data until the end of the
range if given, until a new PLAY request is received, until a PAUSE
request () is received, or until the
end of the media is reached. If no Range header is present in the
PLAY request, the server SHALL play from current pause point until
the end of media. The pause point defaults at session start to the
beginning of the media. For media that is time-progressing and has
no retention, the pause point will always be set equal to NPT "now",
i.e., the current delivery point. The pause point may also be set to
a particular point in the media by the PAUSE method; see . The pause point for media that is currently
playing is equal to the current media position. For time-progressing
media with time-limited retention, if the pause point represents a
position that is older than what is retained by the server, the
pause point will be moved to the oldest retained position.What range values are valid depends on the type of content. For
content that isn't time-progressing, the range value is valid if the
given range is part of any media within the aggregate. In other
words, the valid media range for the aggregate is the union of all
of the media components in the aggregate. If a given range value
points outside of the media, the response MUST be the 457 (Invalid
Range) error code and include the
Media-Range header with the valid range for the media. Except
for time-progressing content where the client requests a start point
prior to what is retained, the start point is adjusted to the oldest
retained content. For a start point that is beyond the media front
edge, i.e., beyond the current value for "now", the server SHALL
adjust the start value to the current front edge. The Range header's
stop point value may point beyond the current media edge. In that
case, the server SHALL deliver media from the requested (and
possibly adjusted) start point until the first of either the
provided stop point or the end of the media. Please note that if one
simply wants to play from a particular start point until the end of
media, using a Range header with an implicit stop point is
RECOMMENDED.If a client requests to start playing at the end of media, either
explicitly with a Range header or implicitly with a pause point that
is at the end of media, a 457 (Invalid Range) error MUST be sent and
include the Media-Range
header. It is specified below that the Range header also must
be included in the response and that it will carry the pause point
in the media, in the case of the session being in Ready State. Note
that this also applies if the pause point or requested start point
is at the beginning of the media and a Scale header is included with a negative
value (playing backwards).For media with random access properties, a client may indicate
which policy for start point selection the server should use. This
is done by including the Seek-Style
header in the PLAY request. The Seek-Style applied will
affect the content of the Range header as it will be adjusted to
indicate from what point the media actually is delivered.A client desiring to play the media from the beginning MUST send
a PLAY request with a Range header pointing at the beginning, e.g.,
"npt=0-". If a PLAY request is received without a Range header and
media delivery has stopped at the end, the server SHOULD respond
with a 457 (Invalid Range) error response. In that response, the
current pause point MUST be included in a Range header.All range specifiers in this specification allow for ranges with
an implicit start point (e.g., "npt=-30"). When used in a PLAY
request, the server treats this as a request to start or resume
delivery from the current pause point, ending at the end time
specified in the Range header. If the pause point is located later
than the given end value, a 457 (Invalid Range) response MUST be
returned.The example below will play seconds 10 through 25. It also
requests that the server deliver media from the first random access
point prior to the indicated start point.Servers MUST include a Range header in any PLAY response, even if
no Range header was present in the request. The response MUST use
the same format as the request's Range header contained. If no Range
header was in the request, the format used in any previous PLAY
request within the session SHOULD be used. If no format has been
indicated in a previous request, the server MAY use any time format
supported by the media and indicated in the Accept-Ranges header in
the SETUP request. It is RECOMMENDED that NPT is used if supported
by the media.For any error response to a PLAY request, the server's response
depends on the current session state. If the session is in Ready
state, the current pause point is returned using a Range header with
the pause point as the explicit start point and an implicit stop
point. For time-progressing content, where the pause-point moves
with real-time due to limited retention, the current pause point is
returned. For sessions in Play state, the current playout point and
the remaining parts of the range request are returned. For any media
with retention longer than 0 seconds, the currently valid
Media-Range header SHALL also be included in the response.A PLAY response MAY include a header carrying synchronization
information. As the information necessary is dependent on the
media-transport format, further rules specifying the header and its
usage are needed. For RTP the RTP-Info header is specified, see
, and used in the following
example.Here is a simple example for a single audio stream where the
client requests the media starting from 3.52 seconds and to the end.
The server sends a 200 OK response with the actual play time, which
is 10 ms prior (3.51), and the RTP-Info header that contains the
necessary parameters for the RTP stack.The server replies with the actual start point that will be
delivered. This may differ from the requested range if alignment of
the requested range to valid frame boundaries is required for the
media source. Note that some media streams in an aggregate may need
to be delivered from even earlier points. Also, some media formats
have a very long duration per individual data unit; therefore, it
might be necessary for the client to parse the data unit, and select
where to start. The server SHALL also indicate which policy it uses
for selecting the actual start point by including a Seek-Style
header.In the following example, the client receives the first media
packet that stretches all the way up and past the requested
playtime. Thus, it is the client's decision whether to render to the
user the time between 3.52 and 7.05 or to skip it. In most cases, it
is probably most suitable not to render that time period.After playing the desired range, the presentation does NOT change
to the Ready state, media delivery simply stops. If it is necessary
to put the stream into the Ready state, a PAUSE request MUST be
issued. A PLAY request while the stream is still in the Play state
is legal and can be issued without an intervening PAUSE request.
Such a request MUST replace the current PLAY action with the new one
requested, i.e., being handled in the same way as if as the request
was received in Ready state. In the case that the range in the Range
header has an implicit start time ("-endtime"), the server MUST
continue to play from where it currently was until the specified
endpoint. This is useful to change the end to at another point than
in the previous request.The following example plays the whole presentation starting at
SMPTE time code 0:10:20 until the end of the clip. Note: the
RTP-Info headers have been broken into several lines, where
subsequent lines start with whitespace as allowed by the syntax.For playing back a recording of a live presentation, it may be
desirable to use clock units:PLAY requests can operate on sessions controlling a single media
stream and on aggregated sessions controlling multiple media
streams.In an aggregated session, the PLAY request MUST contain an
aggregated control URI. A server MUST respond with a 460 error (Only
Aggregate Operation Allowed) if the client PLAY Request-URI is for a
single media. The media in an aggregate MUST be played in sync. If a
client wants individual control of the media, it needs to use
separate RTSP sessions for each media.For aggregated sessions where the initial SETUP request (creating
a session) is followed by one or more additional SETUP requests, a
PLAY request MAY be pipelined after
those additional SETUP requests without awaiting their responses.
This procedure can reduce the delay from the start of session
establishment until media playout has started with one RTT. However,
a client needs to be aware that using this procedure will result in
the playout of the server state established at the time of
processing the PLAY, i.e., after the processing of all the requests
prior to the PLAY request in the pipeline. This state may not be the
intended one due to failure of any of the prior requests. A client
can easily determine this based on the responses from those
requests. In case of failure, the client can halt the media playout
using PAUSE and try to establish the intended state again before
issuing another PLAY request.Clients can issue PLAY requests while the stream is in Play state
and thus updating their request.The important difference compared to a PLAY request in Ready
state is the handling of the current play point and how the Range
header in the request is constructed. The session is actively
playing media and the play point will be moving, making the exact
time a request will take effect hard to predict. Depending on how
the PLAY header appears, two different cases exist: total
replacement or continuation. A total replacement is signaled by
having the first range specification have an explicit start value,
e.g., "npt=45-" or "npt=45-60", in which case the server stops
playout at the current playout point and then starts delivering
media according to the Range header. This is equivalent to having
the client first send a PAUSE and then a new PLAY request that isn't
based on the pause point. In the case of continuation, the first
range specifier has an implicit start point and an explicit stop
value (Z), e.g., "npt=-60", which indicate that it MUST convert the
range specifier being played prior to this PLAY request (X to Y)
into (X to Z) and continue as if this was the request originally
played. If the current delivery point is beyond the stop point, the
server SHALL immediately pause delivery. As the request has been
completed successfully, it shall be responded to with a 200 OK
response. A PLAY_NOTIFY with end-of-stream is also sent to indicate
the actual stop point. The pause point is set to the requested stop
point.The following is an example of this behavior: The server has
received requests to play ranges 10 to 15. If the new PLAY request
arrives at the server 4 seconds after the previous one, it will take
effect while the server still plays the first range (10-15). The
server changes the current play to continue to 25 seconds, i.e., the
equivalent single request would be PLAY with "range: npt=10-25".A common use of a PLAY request while in Play state is changing
the scale of the media, i.e., entering or leaving fast forward or
fast rewind. The client can issue an updating PLAY request that is
either a continuation or a complete replacement, as discussed above
this section. Below is an example of a client that is requesting a
fast forward (scale = 2) without giving a stop point and then a
change from fast forward to regular playout (scale = 1). In the
second PLAY request, the time is set explicitly to be wherever the
server currently plays out (npt=now-) and the server responds with
the actual playback point where the new scale actually takes effect
(npt=02:17:27.144-).On-demand media is indicated by the content of the
Media-Properties header in the SETUP response when (see also ):the Random Access property is set to Random-Access;the Content Modifications property is set to Immutable;the Retention property is set to Unlimited or
Time-Limited.Playing on-demand media follows the general usage as
described in .Dynamic on-demand media is indicated by the content of the
Media-Properties header in the SETUP response when (see also ):the Random Access property is set to Random-Access;the Content Modifications property is set to Dynamic;the Retention property is set to Unlimited or
Time-Limited.Playing on-demand media follows the general usage as described in
as long as the media has not been
changed.There are two ways for the client to be informed about changes of
media resources in Play state. The first being that the client will
receive a PLAY_NOTIFY request with the Notify-Reason header set to
media-properties-update (see ). The client can use
the value of the Media-Range header to decide further actions, if
the Media-Range header is present in the PLAY_NOTIFY request. The
second way is that the client issues a GET_PARAMETER request without
a body but including a Media-Range header. The 200 OK response MUST
include the current Media-Range header (see ).Live media is indicated by the content of the Media-Properties
header in the SETUP response when (see also ):the Random Access property is set to No-Seeking;the Content Modifications property is set to
Time-Progressing;the Retention property's Time-Duration is set to 0.0.For live media, the SETUP 200 OK response MUST include the
Media-Range header (see ).A client MAY send PLAY requests without the Range header. If the
request includes the Range header, it MUST use a symbolic value
representing "now". For NPT, that range specification is "npt=now-".
The server MUST include the Range header in the response, and it
MUST indicate an explicit time value and not a symbolic value. In
other words, "npt=now-" cannot be used in the response. Instead, the
time since session start is recommended, expressed as an open
interval, e.g., "npt=96.23-". An absolute time value (clock) for the
corresponding time MAY be given, i.e., "clock=20030213T143205Z-".
The Absolute Time format can only be used if the client has shown
support for it using the Accept-Ranges header.Certain media servers may offer recording services of live
sessions to their clients. This recording would normally be from the
beginning of the media session. Clients can randomly access the
media between now and the beginning of the media session. This live
media with recording is indicated by the content of the
Media-Properties header in the SETUP response when (see also ):the Random Access property is set to Random-Access;the Content Modifications property is set to
Time-Progressing;the Retention property is set to Time-Limited or
UnlimitedThe SETUP 200 OK response MUST include the Media-Range header
(see ) for this type of media. For
live media with recording, the Range header indicates the current
delivery point in the media and the Media-Range header indicates the
currently available media window around the current time. This
window can cover recorded content in the past (seen from current
time in the media) or recorded content in the future (seen from
current time in the media). The server adjusts the delivery point to
the requested border of the window. If the client requests a
delivery point that is located outside the recording window, e.g.,
if the requested point is too far in the past, the server selects
the oldest point in the recording. The considerations in apply if a client requests delivery with
scale values other than 1.0 (normal
playback rate) while delivering live media with recording.Certain media servers may offer time-shift services to their
clients. This time shift records a fixed interval in the past, i.e.,
a sliding window recording mechanism, but not past this interval.
Clients can randomly access the media between now and the interval.
This live media with recording is indicated by the content of the
Media-Properties header in the SETUP response when (see also ):the Random Access property is set to Random-Access;the Content Modifications property is set to
Time-Progressing;the Retention property is set to Time-Duration and a value
indicating the recording interval (>0).The SETUP 200 OK response MUST include the Media-Range header
(see ) for this type of media. For
live media with recording, the Range header indicates the current
time in the media and the Media-Range header indicates a window
around the current time. This window can cover recorded content in
the past (seen from current time in the media) or recorded content
in the future (seen from current time in the media). The server
adjusts the play point to the requested border of the window, if the
client requests a play point that is located outside the recording
windows, e.g., if requested too far in the past, the server selects
the oldest range in the recording. The considerations in apply if a client requests delivery using
a scale value other than 1.0 (normal
playback rate) while delivering live media with time-shift.The PLAY_NOTIFY method is issued by a server to inform a client
about an asynchronous event for a session in Play state. The Session
header MUST be presented in a PLAY_NOTIFY request and indicates the
scope of the request. Sending of PLAY_NOTIFY requests requires a
persistent connection between server and client; otherwise, there is
no way for the server to send this request method to the client.PLAY_NOTIFY requests have an end-to-end (i.e., server-to-client)
scope, as they carry the Session header, and apply only to the given
session. The client SHOULD immediately return a response to the
server.PLAY_NOTIFY requests MAY use both an aggregate control URI and
individual media resource URIs, depending on the scope of the
notification. This scope may have important distinctions for
aggregated sessions, and each reason for a PLAY_NOTIFY request needs
to specify the interpretation as well as if aggregated control URIs or
individual URIs may be used in requests.PLAY_NOTIFY requests can be used with a message body, depending on
the value of the Notify-Reason header. It is described in the
particular section for each Notify-Reason if a message body is used.
However, currently there is no Notify-Reason that allows the use of a
message body. In this case, there is a need to obey some limitations
when adding new Notify-Reasons that intend to use a message body: the
server can send any type of message body, but it is not ensured that
the client can understand the received message body. This is related
to DESCRIBE (see ); but, in this
particular case, the client can state its acceptable message bodies by
using the Accept header. In the case of PLAY_NOTIFY, the server does
not know which message bodies are understood by the client.The Notify-Reason header (see )
specifies the reason why the server sends the PLAY_NOTIFY request.
This is extensible and new reasons can be added in the future (see
). In case the client
does not understand the reason for the notification, it MUST respond
with a 465 (Notification Reason
Unknown) error code. This document defines how servers can send
PLAY_NOTIFY with Notify-Reason values of these types:end-of-stream (see );media-properties-update (see );scale-change (see ).A PLAY_NOTIFY request with the Notify-Reason header set to
end-of-stream indicates the completion or near completion of the
PLAY request and the ending delivery of the media stream(s). The
request MUST NOT be issued unless the server is in the Play state.
The end of the media stream delivery notification may be used either
to indicate a successful completion of the PLAY request currently
being served or to indicate some error resulting in failure to
complete the request. The Request-Status header MUST be
included to indicate which request the notification is for and its
completion status. The message
response status codes are used to indicate how the PLAY
request concluded. The sender of a PLAY_NOTIFY MAY issue an updated
PLAY_NOTIFY, in the case of a PLAY_NOTIFY sent with wrong
information. For instance, a PLAY_NOTIFY was issued before reaching
the end-of-stream, but some error occurred resulting in that the
previously sent PLAY_NOTIFY contained a wrong time when the stream
will end. In this case, a new PLAY_NOTIFY MUST be sent including the
correct status for the completion and all additional
information.PLAY_NOTIFY requests with the Notify-Reason header set to
end-of-stream MUST include a Range header and the Scale header if
the scale value is not 1. The Range header indicates the point in
the stream or streams where delivery is ending with the timescale
that was used by the server in the PLAY response for the request
being fulfilled. The server MUST NOT use the "now" constant in the
Range header; it MUST use the actual numeric end position in the
proper timescale. When end-of-stream notifications are issued prior
to having sent the last media packets, this is made evident because
the end time in the Range header is beyond the current time in the
media being received by the client, e.g., "npt=-15", if npt is
currently at 14.2 seconds. The Scale header is to be included so
that it is evident if the media timescale is moving backwards or has
a non-default pace. The end-of-stream notification does not prevent
the client from sending a new PLAY request.If RTP is used as media transport, an RTP-Info header MUST be
included, and the RTP-Info header MUST indicate the last sequence
number in the sequence parameter.For an RTSP Session where media resources are under aggregated
control, the media resources will normally end at approximately the
same time, although some small differences may exist, on the scale
of a few hundred milliseconds. In those cases, an RTSP session under
aggregated control SHOULD send only a single PLAY_NOTIFY request. By
using the aggregate control URI in the PLAY_NOTIFY request, the RTSP
server indicates that this applies to all media resources within the
session. In cases in which RTP is used for media delivery,
corresponding RTP-Info needs to be included for all media resources.
In cases where one or more media resources have a significantly
shorter duration than some other resources in the aggregated
session, the server MAY send end-of-stream notifications using
individual media resource URIs to indicate to agents that there will
be no more media for this particular media resource related to the
current active PLAY request. In such cases, when the remaining media
resources come to the end of the stream, they MUST send a
PLAY_NOTIFY request using the aggregate control URI to indicate that
no more resources remain.A PLAY_NOTIFY request with Notify-Reason header set to
end-of-stream MUST NOT carry a message body.This example request notifies the client about a future
end-of-stream event:A PLAY_NOTIFY request with a Notify-Reason header set to
media-properties-update indicates an update of the media properties
for the given session (see ) or
the available media range that can be played as indicated by the Media-Range header. PLAY_NOTIFY
requests with Notify-Reason header set to media-properties-update
MUST include a Media-Properties and Date header and SHOULD include a
Media-Range header. The Media-Properties header has session scope;
thus, for aggregated sessions, the PLAY_NOTIFY request MUST use the
aggregated control URI.This notification MUST be sent for media that are
time-progressing every time an event happens that changes the basis
for making estimates on how the available for play-back media range
will progress with wall clock time. In addition, it is RECOMMENDED
that the server send these notifications approximately every 5
minutes for time-progressing content to ensure the long-term
stability of the client estimation and allow for clock skew
detection by the client. The time between notifications should be
greater than 1 minute and less than 2 hours. For the reasons just
explained, requests MUST include a Media-Range header to provide
current Media duration and a Range header to indicate the current
playing point and any remaining parts of the requested range.The recommendation for sending updates every 5 minutes is due
to any clock skew issues. In 5 minutes, the clock skew should
not become too significant as this is not used for media
playback and synchronization, it is only for determining which
content is available to the user.A PLAY_NOTIFY request with Notify-Reason header set to
media-properties-update MUST NOT carry a message body.The server may be forced to change the rate of media time per
playback time when a client requests delivery using a scale value other than 1.0 (normal
playback rate). For time-progressing media with some retention,
i.e., the server stores already-sent content, a client requesting to
play with scale values larger than 1 may catch up with the front end
of the media. The server will then be unable to continue to provide
content at scale larger than 1 as content is only made available by
the server at scale = 1. Another case is when scale < 1 and the
media retention is Time-Duration limited. In this case, the delivery
point can reach the oldest media unit available, and further
playback at this scale becomes impossible as there will be no media
available. To avoid having the client lose any media, the scale will
need to be adjusted to the same rate at which the media is removed
from the storage buffer, commonly scale = 1.0.Another case is when the content itself consists of spliced
pieces or is dynamically updated. In these cases, the server may be
required to change from one supported scale value (different than
scale = 1.0) to another. In this case, the server will pick the
closest value and inform the client of what it has picked. In these
cases, the media properties will also be sent, updating the
supported scale values. This enables a client to adjust the scale
value used.To minimize impact on playback in any of the above cases, the
server MUST modify the playback properties, set scale to a
supportable value, and continue delivery of the media. When doing
this modification, it MUST send a PLAY_NOTIFY message with the
Notify-Reason header set to "scale-change". The request MUST contain
a Range header with the media time when the change took effect, a
Scale header with the new value in use, a Session header with the
identifier for the session to which it applies, and a Date header
with the server wallclock time of the change. For time-progressing
content, the Media-Range and the Media-Properties headers at this
point in time also MUST be included. The Media-Properties header
MUST be included if the scale change was due to the content changing
what scale values ("Scales") are supported.For media streams delivered using RTP, an RTP-Info header MUST
also be included. It MUST contain the rtptime parameter with a value
corresponding to the point of change in that media and optionally
the sequence number.PLAY_NOTIFY requests for aggregated sessions MUST use the
aggregated control URI in the request. The scale change for any
aggregated session applies to all media streams that are part of the
aggregate.A PLAY_NOTIFY request with Notify-Reason header set to
"Scale-Change" MUST NOT carry a message body.The PAUSE request causes the stream delivery to immediately be
interrupted (halted). A PAUSE request MUST be made either with the
aggregated control URI for aggregated sessions, resulting in all media
being halted, or with the media URI for non-aggregated sessions. Any
attempt to mute a single media with a PAUSE request in an aggregated
session MUST be responded to with a 460 (Only Aggregate Operation
Allowed) error. After resuming playback, synchronization of the tracks
MUST be maintained. Any server resources are kept, though servers MAY
close the session and free resources after being paused for the
duration specified with the timeout parameter of the Session header in
the SETUP message.Example:The PAUSE request causes stream delivery to be interrupted
immediately on receipt of the message, and the pause point is set to
the current point in the presentation. That pause point in the media
stream needs to be maintained. A subsequent PLAY request without a
Range header resumes from the pause point and plays until media
end.The pause point after any PAUSE request MUST be returned to the
client by adding a Range header with what remains unplayed of the PLAY
request's range. For media with random access properties, if one
desires to resume playing a ranged request, one simply includes the
Range header from the PAUSE response and includes the Seek-Style
header with the Next policy in the PLAY request. For media that is
time-progressing and has retention duration=0, the follow-up PLAY
request to start media delivery again MUST use "npt=now-" and not the
answer given in the response to PAUSE.If a client issues a PAUSE request and the server acknowledges and
enters the Ready state, the proper server response, if the player
issues another PAUSE, is still 200 OK. The 200 OK response MUST
include the Range header with the current pause point. See examples
below:The TEARDOWN client-to-server request stops the stream delivery
for the given URI, freeing the resources associated with it. A
TEARDOWN request can be performed on either an aggregated or a media
control URI. However, some restrictions apply depending on the
current state. The TEARDOWN request MUST contain a Session header
indicating to what session the request applies. The TEARDOWN request
MUST NOT include a Terminate-Reason header.A TEARDOWN using the aggregated control URI or the media URI in a
session under non-aggregated control (single media session) MAY be
done in any state (Ready and Play). A successful request MUST result
in that media delivery being immediately halted and the session
state being destroyed. This MUST be indicated through the lack of a
Session header in the response.A TEARDOWN using a media URI in an aggregated session can only be
done in Ready state. Such a request only removes the indicated media
stream and associated resources from the session. This may result in
a session returning to non-aggregated control, because it only
contains a single media after the request's completion. A session
that will exist after the processing of the TEARDOWN request MUST,
in the response to that TEARDOWN request, contain a Session header.
Thus, the presence of the Session header indicates to the receiver
of the response if the session is still extant or has been
removed.Example:The server can send TEARDOWN requests in the server-to-client
direction to indicate that the server has been forced to terminate
the ongoing session. This may happen for several reasons, such as
server maintenance without available backup, or that the session has
been inactive for extended periods of time. The reason is provided
in the Terminate-Reason
header.When an RTSP client has maintained an RTSP session that otherwise
is inactive for an extended period of time, the server may reclaim
the resources. That is done by issuing a TEARDOWN request with the
Terminate-Reason set to "Session-Timeout". This MAY be done when the
client has been inactive in the RTSP session for more than one Session Timeout period. However, the
server is NOT RECOMMENDED to perform this operation until an
extended period of inactivity of 10 times the Session-Timeout period
has passed. It is up to the operator of the RTSP server to actually
configure how long this extended period of inactivity is. An
operator should take into account, when doing this configuration,
what the served content is and what this means for the extended
period of inactivity.In case the server needs to stop providing service to the
established sessions and there is no server to point at in a
REDIRECT request, then TEARDOWN SHALL be used to terminate the
session. This method can also be used when non-recoverable internal
errors have happened and the server has no other option than to
terminate the sessions.The TEARDOWN request MUST be made only on the session aggregate
control URI (i.e., it is not allowed to terminate individual media
streams, if it is a session aggregate), and it MUST include the
following headers: Session and Terminate-Reason. The request only
applies to the session identified in the Session header. The server
may include a message to the client's user with the "user-msg"
parameter.The TEARDOWN request may alternatively be done on the wildcard
URI "*" and without any session header. The scope of such a request
is limited to the next-hop (i.e., the RTSP agent in direct
communication with the server) and applies, as well, to the RTSP
connection between the next-hop RTSP agent and the server. This
request indicates that all sessions and pending requests being
managed via the connection are terminated. Any intervening proxies
SHOULD do all of the following in the order listed: respond to the TEARDOWN requestdisconnect the control channel from the requesting serverpass the TEARDOWN request to each applicable client
(typically those clients with an active session or an unanswered
request)Note: The proxy is responsible for accepting TEARDOWN
responses from its clients; these responses MUST NOT be passed
on to either the original server or the target server in the
redirect.The GET_PARAMETER request retrieves the value of any specified
parameter or parameters for a presentation or stream specified in the
URI. If the Session header is present in a request, the value of a
parameter MUST be retrieved in the specified session context. There
are two ways of specifying the parameters to be retrieved.The first approach includes headers that have been defined to be
usable for this purpose. Headers for this purpose should allow empty,
or stripped value parts to avoid having to specify bogus data when
indicating the desire to retrieve a value. The successful completion
of the request should also be evident from any filled out values in
the response. The headers in this specification that MAY be used for
retrieving their current value using GET_PARAMETER are listed below;
additional headers MAY be specified in the future:Accept-RangesMedia-RangeMedia-PropertiesRangeRTP-InfoThe other way is to specify a message body that lists the
parameter(s) that are desired to be retrieved. The Content-Type header is used to
specify which format the message body has. If the receiver of the
request does not support the media type used for the message body, it
SHALL respond using the error code 415 (Unsupported Media Type). The
responder to a GET_PARAMETER request MUST use the media type of the
request for the response. For additional considerations regarding
message body negotiation, see .RTSP agents implementing support for responding to GET_PARAMETER
requests SHALL implement the "text/parameters" format. This to
ensure that at least one known format for parameters is implemented
and, thus, prevent parameter format negotiation failure.Parameters specified within the body of the message must all be
understood by the request-receiving agent. If one or more parameters
are not understood a 451 (Parameter Not Understood) MUST be sent
including a body listing the parameters that weren't understood. If
all parameters are understood, their values are filled in and returned
in the response message body.The method can also be used without a message body or any header
that requests parameters for keep-alive purposes. The keep-alive timer
has been updated for any request that is successful, i.e., a 200 OK
response is received. Any non-required header present in such a
request may or may not have been processed. Normally, the presence of
filled-out values in the header will be indication that the header has
been processed. However, for cases when this is difficult to
determine, it is recommended to use a feature tag and the Require
header. For this reason, it is usually easier if any parameters to be
retrieved are sent in the body, rather than using any header.Example:This method requests the setting of the value of a parameter or a
set of parameters for a presentation or stream specified by the URI.
If the Session header is present in a request, the value of a
parameter MUST be retrieved in the specified session context. The
method MAY also be used without a message body. It is the RECOMMENDED
method to be used in a request sent for the sole purpose of updating
the keep-alive timer. If this request is successful, i.e., a 200 OK
response is received, then the keep-alive timer has been updated. Any
non-required header present in such a request may or may not have been
processed. To allow a client to determine if any such header has been
processed, it is necessary to use a feature tag and the Require
header. Due to this reason it is RECOMMENDED that any parameters are
sent in the body rather than using any header.When using a message body to list the parameter(s) desired to be
set, the Content-Type header is
used to specify which format the message body has. If the receiver of
the request is not supporting the media type used for the message
body, it SHALL respond using the error code 415 (Unsupported Media
Type). For additional considerations regarding message body
negotiation, see .
The responder to a SET_PARAMETER request MUST use the media type of
the request for the response. For additional considerations regarding
message body negotiation, see .RTSP agents implementing support for responding to SET_PARAMETER
requests SHALL implement the text/parameters format. This is to
ensure that at least one known format for parameters is implemented
and, thus, prevent parameter format negotiation failure.A request is RECOMMENDED to only contain a single parameter to
allow the client to determine why a particular request failed. If the
request contains several parameters, the server MUST only act on the
request if all of the parameters can be set successfully. A server
MUST allow a parameter to be set repeatedly to the same value, but it
MAY disallow changing parameter values. If the receiver of the request
does not understand or cannot locate a parameter, error 451 (Parameter
Not Understood) MUST be used. When a parameter is not allowed to
change, the error code is 458 (Parameter Is Read-Only). The response
body MUST contain only the parameters that have errors. Otherwise, a
body MUST NOT be returned. The response body MUST use the media type
of the request for the response.Note: transport parameters for the media stream MUST only be set
with the SETUP command.Restricting setting transport parameters to SETUP is for the
benefit of firewalls connected to border RTSP proxies.The parameters are split in a fine-grained fashion so that
there can be more meaningful error indications. However, it may
make sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client does
not want the camera to pan unless it can also tilt to the right
angle at the same time.Example:The REDIRECT method is issued by a server to inform a client that
the service provided will be terminated and where a corresponding
service can be provided instead. This may happen for different
reasons. One is that the server is being administered such that it
must stop providing service. Thus, the client is required to connect
to another server location to access the resource indicated by the
Request-URI.The REDIRECT request SHALL contain a Terminate-Reason header to inform
the client of the reason for the request. Additional parameters
related to the reason may also be included. The intention here is to
allow a server administrator to do a controlled shutdown of the RTSP
server. That requires sufficient time to inform all entities having
associated state with the server and for them to perform a controlled
migration from this server to a fall-back server.A REDIRECT request with a Session header has end-to-end (i.e.,
server-to-client) scope and applies only to the given session. Any
intervening proxies SHOULD NOT disconnect the control channel while
there are other remaining end-to-end sessions. The REQUIRED Location
header MUST contain a complete absolute URI pointing to the resource
to which the client SHOULD reconnect. Specifically, the Location MUST
NOT contain just the host and port. A client may receive a REDIRECT
request with a Session header, if and only if, an end-to-end session
has been established.A client may receive a REDIRECT request without a Session header at
any time when it has communication or a connection established with a
server. The scope of such a request is limited to the next-hop (i.e.,
the RTSP agent in direct communication with the server) and applies to
all sessions controlled, as well as the connection between the
next-hop RTSP agent and the server. A REDIRECT request without a
Session header indicates that all sessions and pending requests being
managed via the connection MUST be redirected. The Location header, if
included in such a request, SHOULD contain an absolute URI with only
the host address and the OPTIONAL port number of the server to which
the RTSP agent SHOULD reconnect. Any intervening proxies SHOULD do all
of the following in the order listed: respond to the REDIRECT requestdisconnect the control channel from the requesting serverconnect to the server at the given host addresspass the REDIRECT request to each applicable client (typically
those clients with an active session or an unanswered request)Note: The proxy is responsible for accepting REDIRECT responses
from its clients; these responses MUST NOT be passed on to either
the original server or the redirected server.A server that needs to terminate a session or all its sessions and
lacks an alternative server to redirect to, SHALL instead use TEARDOWN
requests.When no Terminate-Reason "time" parameter is included in a REDIRECT
request, the client SHALL perform the redirection immediately and
return a response to the server. The server shall consider the session
to be terminated and can free any associated state after it receives
the successful (2xx) response. The server MAY close the signaling
connection upon receiving the response, and the client SHOULD close
the signaling connection after sending the 2xx response. The exception
to this is when the client has several sessions on the server being
managed by the given signaling connection. In this case, the client
SHOULD close the connection when it has received and responded to
REDIRECT requests for all the sessions managed by the signaling
connection.The Terminate-Reason header "time" parameter MAY be used to
indicate the wallclock time by which the redirection MUST have taken
place. To allow a client to determine that redirect time without being
time synchronized with the server, the server MUST include a Date
header in the request. The client should have terminated the session
and closed the connection before the redirection time-line terminated.
The server MAY simply cease to provide service when the deadline time
has been reached, or it can issue a TEARDOWN requests to the remaining
sessions.If the REDIRECT request times out following the rules in , the server MAY terminate the
session or transport connection that would be redirected by the
request. This is a safeguard against misbehaving clients that refuse
to respond to a REDIRECT request. This action removes any incentive of
not acknowledging the reception of a REDIRECT request.After a REDIRECT request has been processed, a client that wants to
continue to receive media for the resource identified by the
Request-URI will have to establish a new session with the designated
host. If the URI given in the Location header is a valid resource URI,
a client SHOULD issue a DESCRIBE request for the URI.Note: The media resource indicated by the Location header can
be identical, slightly different, or totally different. This is
the reason why a new DESCRIBE request SHOULD be issued.If the Location header contains only a host address, the client may
assume that the media on the new server is identical to the media on
the old server, i.e., all media configuration information from the old
session is still valid except for the host address. However, the usage
of conditional SETUP using MTag identifiers is RECOMMENDED as a means
to verify the assumption.This example request redirects traffic for this session to the new
server at the given absolute time:In order to fulfill certain requirements on the network side, e.g.,
in conjunction with network address translators that block RTP traffic
over UDP, it may be necessary to interleave RTSP messages and
media-stream data. This interleaving should generally be avoided unless
necessary since it complicates client and server operation and imposes
additional overhead. Also, head-of-line blocking may cause problems.
Interleaved binary data SHOULD only be used if RTSP is carried over TCP.
Interleaved data is not allowed inside RTSP messages.Stream data, such as RTP packets, is encapsulated by an ASCII dollar
sign (36 decimal) followed by a one-octet channel identifier and the
length of the encapsulated binary data as a binary, two-octet unsigned
integer in network octet order (Appendix B of ).
The stream data follows immediately afterwards, without a CRLF, but
including the upper-layer protocol headers. Each dollar sign block MUST
contain exactly one upper-layer protocol data unit, e.g., one RTP
packet. Note that this mechanism does not support PDUs larger than 65535
octets, which matches the maximum payload size of regular, non-jumbo
IPv4 and IPv6 packets. If the media delivery protocol intended to be
used has larger PDUs than that, a definition of a PDU fragmentation
mechanism will be required to support embedded binary data.The channel identifier is defined in the Transport header with the
interleaved parameter ().When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. The usage of RTCP messages is
indicated by including an interval containing a second channel in the
interleaved parameter of the Transport header (see ). If RTCP is used, packets MUST be sent on the
first available channel that is higher than the RTP channel. The
channels are bidirectional, using the same Channel ID in both
directions; therefore, RTCP traffic is sent on the second channel in
both directions.RTCP is sometimes needed for synchronization when two or more
streams are interleaved in such a fashion. Also, this provides a
convenient way to tunnel RTP/RTCP packets through the RTSP
connection (TCP or TCP/TLS) when required by the network
configuration and to transfer them onto UDP when possible.RTSP Proxies are RTSP agents that are located in between a client and
a server. A proxy can take on the roles of both client and server
depending on what it tries to accomplish. RTSP proxies use two
transport-layer connections: one from the RTSP client to the RTSP proxy
and a second from the RTSP proxy to the RTSP server. Proxies are
introduced for several different reasons; those listed below are often
combined.This type of proxy is used to reduce
the workload on servers and connections. By caching the description
and media streams, i.e., the presentation, the proxy can serve a
client with content, but without requesting it from the server once
it has been cached and has not become stale. See . This type of proxy is also expected to
understand RTSP endpoint functionality, i.e., functionality
identified in the Require header in addition to what Proxy-Require
demands.This type of proxy is used to ensure
that an RTSP client gets access to servers and content on an
external network or gets access by using content encodings not
supported by the client. The proxy performs the necessary
translation of addresses, protocols, or encodings. This type of
proxy is expected also to understand RTSP endpoint functionality,
i.e., functionality identified in the Require header in addition to
what Proxy-Require demands.This type of proxy is used to ensure
that an RTSP client gets access to servers on an external network.
Thus, this proxy is placed on the border between two domains, e.g.,
a private address space and the public Internet. The proxy performs
the necessary translation, usually addresses. This type of proxy is
required to redirect the media to itself or a controlled gateway
that performs the translation before the media can reach the
client.This type of proxy is used to help
facilitate security functions around RTSP. For example, in the case
of a firewalled network, the security proxy requests that the
necessary pinholes in the firewall are opened when a client in the
protected network wants to access media streams on the external
side. This proxy can perform its function without redirecting the
media between the server and client. However, in deployments with
private address spaces, this proxy is likely to be combined with the
access proxy. The functionality of this proxy is usually closely
tied into understanding all aspects of the media transport.RTSP proxies can also provide network
owners with a logging and auditing point for RTSP sessions, e.g.,
for corporations that track their employees usage of the network.
This type of proxy can perform its function without inserting itself
or any other node in the media transport. This proxy type can also
accept unknown methods as it doesn't interfere with the clients'
requests.All types of proxies can also be used when using secured
communication with TLS, as RTSP 2.0 allows the client to approve
certificate chains used for connection establishment from a proxy; see
. However, that trust model may
not be suitable for all types of deployment. In those cases, the secured
sessions do bypass the proxies.Access proxies SHOULD NOT be used in equipment like NATs and
firewalls that aren't expected to be regularly maintained, like home or
small office equipment. In these cases, it is better to use the NAT
traversal procedures defined for RTSP 2.0 . The
reason for these recommendations is that any extensions of RTSP
resulting in new media-transport protocols or profiles, new parameters,
etc., may fail in a proxy that isn't maintained. This would impede
RTSP's future development and usage.The existence of proxies must always be considered when developing
new RTSP extensions. Most types of proxies will need to implement any
new method to operate correctly in the presence of that extension. New
headers can be introduced and will not be blocked by older proxies.
However, it is important to consider if this header and its function
are required to be understood by the proxy or if it can be simply
forwarded. If the header needs to be understood, a feature tag
representing the functionality MUST be included in the Proxy-Require
header. Below are guidelines for analysis whether the header needs to
be understood. The Transport header and its parameters are extensible,
which requires handling rules for a proxy in order to ensure a correct
interpretation.Whether or not a proxy needs to understand a header is not easy to
determine as they serve a broad variety of functions. When evaluating
if a header needs to be understood, one can divide the functionality
into three main categories:The caching and translator proxies
modify the actual media and therefore need also to understand the
request directed to the server that affects how the media is
rendered. Thus, this type of proxy also needs to understand the
server-side functionality.The access and the security
proxy both need to understand how the media transport is
performed, either for opening pinholes or translating the outer
headers, e.g., IP and UDP or TCP.The audit proxy is special in that it
does not modify the messages in other ways than to insert the Via
header. That makes it possible for this type to forward RTSP
messages that contain different types of unknown methods, headers,
or header parameters.An extension has to be classified as mandatory to be
implemented for a proxy, if an extension has to be understood by a
"Transport modifying" type of proxy.RTSP proxies may have to multiplex several RTSP sessions from their
clients towards RTSP servers. This requires that RTSP requests from
multiple clients be multiplexed onto a common connection for requests
outgoing to an RTSP server, and, on the way back, the responses be
demultiplexed from the server to per-client responses. On the protocol
level, this requires that request and response messages be handled in
both directions, requiring that there be a mechanism to correlate
which request/response pair exchanged between proxy and server is
mapped to which client (or client request).This multiplexing of requests and demultiplexing of responses is
done by using the CSeq header field. The proxy has to rewrite the CSeq
in requests to the server and responses from the server and remember
which CSeq is mapped to which client. The proxy also needs to ensure
that the order of the message related to each client is maintained.
defines the handling of how requests and
responses are rewritten.In HTTP, request/response pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE. (Since
the responses for anything but DESCRIBE and GET_PARAMETER do not return
any data, caching is not really an issue for these requests.) However,
it is desirable for the continuous media data, typically delivered
out-of-band with respect to RTSP, to be cached, as well as the session
description.On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up to date by issuing
a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is not up
to date, it modifies the SETUP transport parameters as appropriate and
forwards the request to the origin server. Subsequent control commands
such as PLAY or PAUSE then pass the proxy unmodified. The proxy delivers
the continuous media data to the client, while possibly making a local
copy for later reuse. The exact allowed behavior of the cache is given
by the cache-response directives described in . A cache MUST answer any DESCRIBE requests
if it is currently serving the stream to the requester, as it is
possible that low-level details of the stream description may have
changed on the origin server.Note that an RTSP cache is of the "cut-through" variety. Rather than
retrieving the whole resource from the origin server, the cache simply
copies the streaming data as it passes by on its way to the client.
Thus, it does not introduce additional latency.To the client, an RTSP proxy cache appears like a regular media
server. To the media origin server, an RTSP proxy cache appears like a
client. Just as an HTTP cache has to store the content type, content
language, and so on for the objects it caches, a media cache has to
store the presentation description. Typically, a cache eliminates all
transport references (e.g., multicast information) from the presentation
description, since these are independent of the data delivery from the
cache to the client. Information on the encodings remains the same. If
the cache is able to translate the cached media data, it would create a
new presentation description with all the encoding possibilities it can
offer.When a cache has a stale entry that it would like to use as a
response to a client's request, it first has to check with the origin
server (or possibly an intermediate cache with a fresh response) to
see if its cached entry is still usable. This is called "validating"
the cache entry. To avoid having to pay the overhead of retransmitting
the full response if the cached entry is good, and at the same time
avoiding having to pay the overhead of an extra round trip if the
cached entry is invalid, RTSP supports the use of conditional
methods.The key protocol features for supporting conditional methods are
those concerned with "cache validators." When an origin server
generates a full response, it attaches some sort of validator to it,
which is kept with the cache entry. When a client (user agent or proxy
cache) makes a conditional request for a resource for which it has a
cache entry, it includes the associated validator in the request.The server then checks that validator against the current validator
for the requested resource, and, if they match (see ), it responds with a special
status code (usually, 304 (Not Modified)) and no message body.
Otherwise, it returns a full response (including message body). Thus,
avoiding transmitting the full response if the validator matches and
avoiding an extra round trip if it does not match.In RTSP, a conditional request looks exactly the same as a normal
request for the same resource, except that it carries a special header
(which includes the validator) that implicitly turns the method
(usually DESCRIBE or SETUP) into a conditional.The protocol includes both positive and negative senses of
cache-validating conditions. That is, it is possible to request that a
method be performed either if and only if a validator matches or if
and only if no validators match.Note: a response that lacks a validator may still be cached,
and served from cache until it expires, unless this is explicitly
prohibited by a cache directive (see ). However, a cache cannot perform a
conditional retrieval if it does not have a validator for the
resource, which means it will not be refreshable after it
expires.Media streams that are being adapted based on the transport
capacity between the server and the cache make caching more difficult.
A server needs to consider how it views the caching of media streams
that it adapts and potentially instruct any caches not to cache such
streams.The Last-Modified header ()
value is often used as a cache validator. In simple terms, a cache
entry is considered to be valid if the cache entry was created after
the Last-Modified time.The MTag response-header field-value, a message body tag,
provides for an "opaque" cache validator. This might allow more
reliable validation in situations where it is inconvenient to store
modification dates, where the one-second resolution of RTSP-date
values is not sufficient, or where the origin server wishes to avoid
certain paradoxes that might arise from the use of modification
dates.Message body tags are described in Since both origin servers and caches will compare two validators
to decide if they represent the same or different entities, one
normally would expect that if the message body (i.e., the
presentation description) or any associated message body headers
changes in any way, then the associated validator would change as
well. If this is true, then this validator is a "strong validator".
The Message body (i.e., the presentation description) or any
associated message body headers is named an entity for a better
understanding.However, there might be cases when a server prefers to change the
validator only on semantically significant changes and not when
insignificant aspects of the entity change. A validator that does
not always change when the resource changes is a "weak
validator".Message body tags are normally strong validators, but the
protocol provides a mechanism to tag a message body tag as "weak".
One can think of a strong validator as one that changes whenever the
bits of an entity changes, while a weak value changes whenever the
meaning of an entity changes. Alternatively, one can think of a
strong validator as part of an identifier for a specific entity,
while a weak validator is part of an identifier for a set of
semantically equivalent entities.Note: One example of a strong validator is an integer that is
incremented in stable storage every time an entity is
changed.An entity's modification time, if represented with one-second
resolution, could be a weak validator, since it is possible that
the resource might be modified twice during a single second.Support for weak validators is optional. However, weak
validators allow for more efficient caching of equivalent
objects.A "use" of a validator is either when a client generates a
request and includes the validator in a validating header field or
when a server compares two validators.Strong validators are usable in any context. Weak validators are
only usable in contexts that do not depend on exact equality of an
entity. For example, either kind is usable for a conditional
DESCRIBE of a full entity. However, only a strong validator is
usable for a subrange retrieval, since otherwise the client might
end up with an internally inconsistent entity.Clients MAY issue DESCRIBE requests with either weak or strong
validators. Clients MUST NOT use weak validators in other forms of
requests.The only function that RTSP defines on validators is comparison.
There are two validator comparison functions, depending on whether
or not the comparison context allows the use of weak validators:
The strong comparison function: in order to be considered
equal, both validators MUST be identical in every way, and both
MUST NOT be weak.The weak comparison function: in order to be considered
equal, both validators MUST be identical in every way, but
either or both of them MAY be tagged as "weak" without affecting
the result.A message body tag is strong unless it is explicitly tagged
as weak.A Last-Modified time, when used as a validator in a request, is
implicitly weak unless it is possible to deduce that it is strong,
using the following rules: The validator is being compared by an origin server to the
actual current validator for the entity and,That origin server reliably knows that the associated entity
did not change more than once during the second covered by the
presented validator.ORThe validator is about to be used by a client in an
If-Modified-Since, because the client has a cache entry for the
associated entity, andThat cache entry includes a Date value, which gives the time
when the origin server sent the original response, andThe presented Last-Modified time is at least 60 seconds
before the Date value.ORThe validator is being compared by an intermediate cache to
the validator stored in its cache entry for the entity, andThat cache entry includes a Date value, which gives the time
when the origin server sent the original response, andThe presented Last-Modified time is at least 60 seconds
before the Date value.This method relies on the fact that if two different
responses were sent by the origin server during the same second, but
both had the same Last-Modified time, then at least one of those
responses would have a Date value equal to its Last-Modified time.
The arbitrary 60-second limit guards against the possibility that
the Date and Last-Modified values are generated from different
clocks or at somewhat different times during the preparation of the
response. An implementation MAY use a value larger than 60 seconds,
if it is believed that 60 seconds is too short.If a client wishes to perform a subrange retrieval on a value for
which it has only a Last-Modified time and no opaque validator, it
MAY do this only if the Last-Modified time is strong in the sense
described here.This document adopts a set of rules and recommendations for
origin servers, clients, and caches regarding when various validator
types ought to be used, and for what purposes.RTSP origin servers: SHOULD send a message body tag validator unless it is not
feasible to generate one.MAY send a weak message body tag instead of a strong message
body tag, if performance considerations support the use of weak
message body tags, or if it is unfeasible to send a strong
message body tag.SHOULD send a Last-Modified value if it is feasible to send
one, unless the risk of a breakdown in semantic transparency
that could result from using this date in an If-Modified-Since
header would lead to serious problems.In other words, the preferred behavior for an RTSP origin
server is to send both a strong message body tag and a Last-Modified
value.In order to be legal, a strong message body tag MUST change
whenever the associated entity value changes in any way. A weak
message body tag SHOULD change whenever the associated entity
changes in a semantically significant way.Note: in order to provide semantically transparent caching,
an origin server MUST avoid reusing a specific strong message
body tag value for two different entities or reusing a specific
weak message body tag value for two semantically different
entities. Cache entries might persist for arbitrarily long
periods, regardless of expiration times, so it might be
inappropriate to expect that a cache will never again attempt to
validate an entry using a validator that it obtained at some
point in the past.RTSP clients: If a message body tag has been provided by the origin server,
MUST use that message body tag in any cache-conditional request
(using If-Match or If-None-Match).If only a Last-Modified value has been provided by the origin
server, SHOULD use that value in non-subrange cache-conditional
requests (using If-Modified-Since).If both a message body tag and a Last-Modified value have
been provided by the origin server, SHOULD use both validators
in cache-conditional requests.An RTSP origin server, upon receiving a conditional request
that includes both a Last-Modified date (e.g., in an
If-Modified-Since header) and one or more message body tags (e.g.,
in an If-Match, If-None-Match, or If-Range header field) as cache
validators, MUST NOT return a response status of 304 (Not Modified)
unless doing so is consistent with all of the conditional header
fields in the request.Note: The general principle behind these rules is that RTSP
servers and clients should transmit as much non-redundant
information as is available in their responses and requests.
RTSP systems receiving this information will make the most
conservative assumptions about the validators they receive.The principle behind message body tags is that only the service
author knows the semantics of a resource well enough to select an
appropriate cache validation mechanism, and the specification of any
validator comparison function more complex than octet equality would
open up a can of worms. Thus, comparisons of any other headers are
never used for purposes of validating a cache entry.The effect of certain methods performed on a resource at the origin
server might cause one or more existing cache entries to become
non-transparently invalid. That is, although they might continue to be
"fresh," they do not accurately reflect what the origin server would
return for a new request on that resource.There is no way for RTSP to guarantee that all such cache entries
are marked invalid. For example, the request that caused the change at
the origin server might not have gone through the proxy where a cache
entry is stored. However, several rules help reduce the likelihood of
erroneous behavior.In this section, the phrase "invalidate an entity" means that the
cache will either remove all instances of that entity from its storage
or mark these as "invalid" and in need of a mandatory revalidation
before they can be returned in response to a subsequent request.Some RTSP methods MUST cause a cache to invalidate an entity. This
is either the entity referred to by the Request-URI or by the Location
or Content-Location headers (if present). These methods are: DESCRIBESETUPIn order to prevent DoS attacks, an invalidation based on the
URI in a Location or Content-Location header MUST only be performed if
the host part is the same as in the Request-URI.A cache that passes through requests for methods it does not
understand SHOULD invalidate any entities referred to by the
Request-URI.Where applicable, HTTP status codes (see Section 6 of ) are reused. See in for a listing of which status codes may be
returned by which requests. All error messages, 4xx and 5xx, MAY return
a body containing further information about the error.The requesting agent SHOULD continue with its request. This
interim response is used to inform the requesting agent that the
initial part of the request has been received and has not yet been
rejected by the responding agent. The requesting agent SHOULD
continue by sending the remainder of the request or, if the request
has already been completed, continue to wait for a final response
(see ). The responding agent
MUST send a final response after the request has been completed.This class of status code indicates that the agent's request was
successfully received, understood, and accepted.The request has succeeded. The information returned with the
response is dependent on the method used in the request.The notation "3xx" indicates response codes from 300 to 399
inclusive that are meant for redirection. We use the notation "3rr" to
indicate all 3xx codes used for redirection, i.e., excluding 304. The
304 response code appears here, rather than a 2xx response code, which
would have been appropriate; 304 has also been used in RTSP 1.0.Within RTSP, redirection may be used for load-balancing or
redirecting stream requests to a server topologically closer to the
agent. Mechanisms to determine topological proximity are beyond the
scope of this specification.A 3rr code MAY be used to respond to any request. The Location
header MUST be included in any 3rr response. It is RECOMMENDED that
they are used if necessary before a session is established, i.e., in
response to DESCRIBE or SETUP. However, in cases where a server is not
able to send a REDIRECT request to the agent, the server MAY need to
resort to using 3rr responses to inform an agent with an established
session about the need for redirecting the session. If a 3rr response
is received for a request in relation to an established session, the
agent SHOULD send a TEARDOWN request for the session and MAY
reestablish the session using the resource indicated by the
Location.If the Location header is used in a response, it MUST contain an
absolute URI pointing out the media resource the agent is redirected
to; the URI MUST NOT only contain the hostname.In the event that an unknown 3rr status code is received, the agent
SHOULD behave as if a 302 response code had been received.The 300 response code is not used in RTSP 2.0.The requested resource is moved permanently and resides now at
the URI given by the Location header. The user agent SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message body. The Location header MUST be included in the
response.The requested resource resides temporarily at the URI given by
the Location header. This response is intended to be used for many
types of temporary redirects, e.g., load balancing. It is
RECOMMENDED that the server set the reason phrase to something more
meaningful than "Found" in these cases. The Location header MUST be
included in the response. The user agent SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message body.This example shows a client being redirected to a different
server:This status code MUST NOT be used in RTSP 2.0. However, it was
allowed in RTSP 1.0.If the agent has performed a conditional DESCRIBE or SETUP (see
Sections and
) and the
requested resource has not been modified, the server SHOULD send a
304 response. This response MUST NOT contain a message body.The response MUST include the following header fields: DateMTag or Content-Location, if the headers would have been sent
in a 200 response to the same request.Expires and Cache-Control if the field-value might differ
from that sent in any previous response for the same
variant.This response is independent for the DESCRIBE and SETUP requests.
That is, a 304 response to DESCRIBE does NOT imply that the resource
content is unchanged (only the session description) and a 304
response to SETUP does NOT imply that the resource description is
unchanged. The MTag and If-Match
header may be used to link the DESCRIBE and SETUP in this
manner.The requested resource MUST be accessed through the proxy given
by the Location header that MUST be included. The Location header
field-value gives the URI of the proxy. The recipient is expected to
repeat this single request via the proxy. 305 responses MUST only be
generated by origin servers.The request could not be understood by the agent due to malformed
syntax. The agent SHOULD NOT repeat the request without
modifications. If the request does not have a CSeq header, the agent
MUST NOT include a CSeq in the response.The request requires user authentication using the HTTP authentication mechanism. The usage of
the error code is defined in and any
applicable HTTP authentication scheme, such as Digest. The response is to include a WWW-Authenticate header field
containing a challenge applicable to the requested resource. The
agent can repeat the request with a suitable Authorization header
field. If the request already included authorization credentials,
then the 401 response indicates that authorization has been refused
for those credentials. If the 401 response contains the same
challenge as the prior response, and the user agent has already
attempted authentication at least once, then the user SHOULD be
presented the message body that was given in the response, since
that message body might include relevant diagnostic information.This code is reserved for future use.The agent understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.
If the agent wishes to make public why the request has not been
fulfilled, it SHOULD describe the reason for the refusal in the
message body. If the agent does not wish to make this information
available to the agent, the status code 404 (Not Found) can be used
instead.The agent has not found anything matching the Request-URI. No
indication is given of whether the condition is temporary or
permanent. The 410 (Gone) status code SHOULD be used if the agent
knows, through some internally configurable mechanism, that an old
resource is permanently unavailable and has no forwarding address.
This status code is commonly used when the agent does not wish to
reveal exactly why the request has been refused, or when no other
response is applicable.The method specified in the request is not allowed for the
resource identified by the Request-URI. The response MUST include an
Allow header containing a list of valid methods for the requested
resource. This status code is also to be used if a request attempts
to use a method not indicated during SETUP.The resource identified by the request is only capable of
generating response message bodies that have content characteristics
not acceptable according to the Accept headers sent in the
request.The response SHOULD include a message body containing a list of
available message body characteristics and location(s) from which
the user or user agent can choose the one most appropriate. The
message body format is specified by the media type given in the
Content-Type header field. Depending upon the format and the
capabilities of the user agent, selection of the most appropriate
choice MAY be performed automatically. However, this specification
does not define any standard for such automatic selection.If the response could be unacceptable, a user agent SHOULD
temporarily stop receipt of more data and query the user for a
decision on further actions.This code is similar to 401
(Unauthorized), but it indicates that the client must first
authenticate itself with the proxy. The usage of this error code is
defined in and any applicable HTTP
authentication scheme, such as Digest.
The proxy MUST return a Proxy-Authenticate header
field containing a challenge applicable to the proxy for the
requested resource.The agent did not produce a request within the time that the
agent was prepared to wait. The agent MAY repeat the request without
modifications at any later time.The requested resource is no longer available at the server and
the forwarding address is not known. This condition is expected to
be considered permanent. If the server does not know, or has no
facility to determine, whether or not the condition is permanent,
the status code 404 (Not Found) SHOULD be used instead. This
response is cacheable unless indicated otherwise.The 410 response is primarily intended to assist the task of
repository maintenance by notifying the recipient that the resource
is intentionally unavailable and that the server owners desire that
remote links to that resource be removed. Such an event is common
for limited-time, promotional services and for resources belonging
to individuals no longer working at the server's site. It is not
necessary to mark all permanently unavailable resources as "gone" or
to keep the mark for any length of time -- that is left to the
discretion of the owner of the server.The precondition given in one or more of the 'if-' request-header
fields evaluated to false when it was tested on the agent. See these
sections for the 'if-' headers: If-Match , If-Modified-Since , and If-None-Match . This response code allows the agent to
place preconditions on the current resource meta-information (header
field data) and, thus, prevent the requested method from being
applied to a resource other than the one intended.The agent is refusing to process a request because the request
message body is larger than the agent is willing or able to process.
The agent MAY close the connection to prevent the requesting agent
from continuing the request.If the condition is temporary, the agent SHOULD include a
Retry-After header field to indicate that it is temporary and after
what time the requesting agent MAY try again.The responding agent is refusing to service the request because
the Request-URI is longer than the agent is willing to interpret.
This rare condition is only likely to occur when an agent has used a
request with long query information, when the agent has descended
into a URI "black hole" of redirection (e.g., a redirected URI
prefix that points to a suffix of itself), or when the agent is
under attack by an agent attempting to exploit security holes present
in some agents using fixed-length buffers for reading or
manipulating the Request-URI.The server is refusing to service the request because the message
body of the request is in a format not supported by the requested
resource for the requested method.The recipient of the request does not support one or more
parameters contained in the request. When returning this error
message the agent SHOULD return a message body containing the
offending parameter(s).This status code MUST NOT be used in RTSP 2.0. However, it was
allowed in RTSP 1.0.The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.The RTSP session identifier in the Session header is missing, is
invalid, or has timed out.The agent cannot process this request in its current state. The
response MUST contain an Allow header to make error recovery
possible.The targeted agent could not act on a required request-header.
For example, if PLAY request contains the Range header field but the
stream does not allow seeking. This error message may also be used
for specifying when the time format in Range is impossible for the
resource. In that case, the Accept-Ranges header MUST be returned to
inform the agent of which formats are allowed.The Range value given is out of bounds, e.g., beyond the end of
the presentation.The parameter to be set by SET_PARAMETER can be read but not
modified. When returning this error message, the sender SHOULD
return a message body containing the offending parameter(s).The requested method may not be applied on the URI in question
since it is an aggregate (presentation) URI. The method may be
applied on a media URI.The requested method may not be applied on the URI in question
since it is not an aggregate control (presentation) URI. The method
may be applied on the aggregate control URI.The Transport field did not contain a supported transport
specification.The data transmission channel could not be established because
the agent address could not be reached. This error will most likely
be the result of an agent attempt to place an invalid dest_addr
parameter in the Transport field.The data transmission channel was not established because the
server prohibited access to the agent address. This error is most
likely the result of an agent attempt to redirect media traffic to
another destination with a dest_addr parameter in the Transport
header.The data transmission channel to the media destination is not yet
ready for carrying data. However, the responding agent still expects
that the data transmission channel will be established at some point
in time. Note, however, that this may result in a permanent failure
like 462 (Destination Unreachable).An example of when this error may occur is in the case in which a
client sends a PLAY request to a server prior to ensuring that the
TCP connections negotiated for carrying media data were successfully
established (in violation of this specification). The server would
use this error code to indicate that the requested action could not
be performed due to the failure of completing the connection
establishment.This indicates that the client has received a PLAY_NOTIFY with a Notify-Reason header unknown to
the client.This indicates that there has been an error in a Key Management
function used in conjunction with a request. For example, usage of
Multimedia Internet KEYing (MIKEY)
according to may result in this
error.The secured connection attempt needs user or client authorization
before proceeding. The next hop's certificate is included in this
response in the Accept-Credentials header.When performing a secure connection over multiple connections, an
intermediary has refused to connect to the next hop and carry out
the request due to unacceptable credentials for the used policy.A proxy fails to establish a secure connection to the next-hop
RTSP agent. This is primarily caused by a fatal failure at the TLS
handshake, for example, due to the agent not accepting any cipher
suites.Response status codes beginning with the digit "5" indicate cases
in which the server is aware that it has erred or is incapable of
performing the request. The server SHOULD include a message body
containing an explanation of the error situation and whether it is a
temporary or permanent condition. User agents SHOULD display any
included message body to the user. These response codes are applicable
to any request method.The agent encountered an unexpected condition that prevented it
from fulfilling the request.The agent does not support the functionality required to fulfill
the request. This is the appropriate response when the agent does
not recognize the request method and is not capable of supporting it
for any resource.The agent, while acting as a gateway or proxy, received an
invalid response from the upstream agent it accessed in attempting
to fulfill the request.The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication
is that this is a temporary condition that will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the agent SHOULD
handle the response as it would for a 500 response. The agent MUST
honor the length, if given, in the Retry-After header.Note: The existence of the 503 status code does not imply
that a server must use it when becoming overloaded. Some servers
may wish to simply refuse the transport connection.The response scope is dependent on the request. If the
request was in relation to an existing RTSP session, the scope of
the overload response is to this individual RTSP session. If the
request was not session specific or intended to form an RTSP
session, it applies to the RTSP server identified by the hostname in
the Request-URI.The agent, while acting as a proxy, did not receive a timely
response from the upstream agent specified by the URI or some other
auxiliary server (e.g., DNS) that it needed to access in attempting
to complete the request.The agent does not support, or refuses to support, the RTSP
version that was used in the request message. The agent is
indicating that it is unable or unwilling to complete the request
using the same major version as the agent other than with this error
message. The response SHOULD contain a message body describing why
that version is not supported and what other protocols are supported
by that agent.A feature tag given in the Require or the Proxy-Require fields
was not supported. The Unsupported header MUST be returned stating
the feature for which there is no support.The proxy is currently unable to handle the request due to a
temporary overloading or maintenance of the proxy. The implication
is that this is a temporary condition that will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the agent SHOULD
handle the response as it would for a 500 response. The agent MUST
honor the length, if given in the Retry-After header.Note: The existence of the 553 status code does not imply
that a proxy must use it when becoming overloaded. Some proxies
may wish to simply refuse the connection.The response scope is dependent on the Request. If the
request was in relation to an existing RTSP session, the scope of
the overload response is to this individual RTSP session. If the
request was non-session specific or intended to form an RTSP
session, it applies to all such requests to this proxy.methoddirectionobjectacronymBodyDESCRIBEC -> SP,SDESrGET_PARAMETERC -> S, S -> CP,SGPRR,rOPTIONSC -> S, S -> CP,SOPTPAUSEC -> SP,SPSEPLAYC -> SP,SPLYPLAY_NOTIFYS -> CP,SPNYRREDIRECTS -> CP,SRDRSETUPC -> SSSTPSET_PARAMETERC -> S, S -> CP,SSPRR,rTEARDOWNC -> SP,STRDS -> CPTRDThis table is an overview of RTSP methods, their direction,
and what objects (P: presentation, S: stream) they operate on. "Body"
denotes if a method is allowed to carry body and in which direction; R
= request, r=response. Note: All error messages for statuses 4xx and
5xx are allowed to carry a body.The general syntax for header fields is covered in . This section lists the full set of
header fields along with notes on meaning and usage. The syntax
definitions for header fields are present in . Examples of each header field are
given.Information about header fields in relation to methods and proxy
processing is summarized in Figures , , , and .The "where" column describes the request and response types in which
the header field can be used. Values in this column are: header field may only appear in requests;header field may only appear in responses;numerical value or range indicates
response codes with which the header field can be used;header field is copied from the request to the
response.header field is a general-header and may be present
in both requests and responses.Note: General headers do not always use the "G" value in the "where"
column. This is due to differences when the header may be applied in
requests compared to responses. When such differences exist, they are
expressed using two different rows: one with "where" being "R" and one
with it being "r".The "proxy" column describes the operations a proxy may perform on a
header field. An empty proxy column indicates that the proxy MUST NOT
make any changes to that header, all allowed operations are explicitly
stated: A proxy can add or concatenate the header field if
not present.A proxy can modify an existing header field
value.A proxy can delete a header field-value.A proxy needs to be able to read the header field;
thus, this header field cannot be encrypted.The rest of the columns relate to the presence of a header field in a
method. The method names when abbreviated, are according to : Conditional; requirements on the header field
depend on the context of the message.The header field is mandatory.The header field SHOULD be sent, but agents need
to be prepared to receive messages without that header field.The header field is optional.The header field MUST be present if the message
body is not empty. See Sections , and for details.The header field is not applicable."Optional" means that an agent MAY include the header field in a
request or response. The agent behavior when receiving such headers
varies; for some, it may ignore the header field. In other cases, it is
a request to process the header. This is regulated by the method and
header descriptions. Examples of headers that require processing are the
Require and Proxy-Require header fields discussed in Sections and . A "mandatory" header field MUST be present
in a request, and it MUST be understood by the agent receiving the
request. A mandatory response-header field MUST be present in the
response, and the header field MUST be understood by the processing the
response. "Not applicable" means that the header field MUST NOT be
present in a request. If one is placed in a request by mistake, it MUST
be ignored by the agent receiving the request. Similarly, a header field
labeled "not applicable" for a response means that the agent MUST NOT
place the header field in the response, and the agent MUST ignore the
header field in the response.An RTSP agent MUST ignore extension headers that are not
understood.The From and Location header fields contain a URI. If the URI
contains a comma (') or semicolon (;), the URI MUST be enclosed in
double quotes ("). Any URI parameters are contained within these quotes.
If the URI is not enclosed in double quotes, any semicolon-delimited
parameters are header-parameters, not URI parameters.The Accept request-header field can be used to specify certain
presentation description and parameter media
types that are acceptable for the response to the DESCRIBE
request.See for the syntax.The asterisk "*" character is used to group media types into
ranges, with "*/*" indicating all media types and "type/*" indicating
all subtypes of that type. The range MAY include media type parameters
that are generally applicable to that range.Each media type or range MAY be followed by one or more
accept-params, beginning with the "q" parameter to indicate a relative
quality factor. The first "q" parameter (if any) separates the media
type or range's parameters from the accept-params. Quality factors
allow the user or user agent to indicate the relative degree of
preference for that media type, using the qvalue scale from 0 to 1
(Section 5.3.1 of ). The default value is
q=1.Example of use:Indicates that the requesting agent prefers the media type
application/sdp through the default 1.0 rating but also accepts the
application/example media type with a 0.7 quality rating.If no Accept header field is present, then it is assumed that the
client accepts all media types. If an Accept header field is present,
and if the server cannot send a response that is acceptable according
to the combined Accept field-value, then the server SHOULD send a 406
(Not Acceptable) response.The Accept-Credentials header is a request-header used to indicate
to any trusted intermediary how to handle further secured connections
to proxies or servers. It MUST NOT be included in server-to-client
requests. See for the usage of
this headerIn a request, the header MUST contain the method (User, Proxy, or
Any) for approving credentials selected by the requester. The method
MUST NOT be changed by any proxy, unless it is "Proxy" when a proxy
MAY change it to "user" to take the role of user approving each
further hop. If the method is "User", the header contains zero or more
of the credentials that the client accepts. The header may contain
zero credentials in the first RTSP request to an RTSP server via a
proxy when using the "User" method. This is because the client has not
yet received any credentials to accept. Each credential MUST consist
of one URI identifying the proxy or server, the hash algorithm
identifier, and the hash over that agent's ASN.1 DER-encoded
certificate in Base64,
according to Section 4 of and where the padding bits are set to
zero. All RTSP clients and proxies MUST implement the SHA-256 algorithm for computation of the hash of the
DER-encoded certificate. The SHA-256 algorithm is identified by the
token "sha-256".The intention of allowing for other hash algorithms is to enable
the future retirement of algorithms that are not implemented somewhere
other than here. Thus, the definition of future algorithms for this
purpose is intended to be extremely limited. A feature tag can be used
to ensure that support for the replacement algorithm exists.Example:The Accept-Encoding request-header field is similar to Accept, but
it restricts the content-codings (see ), i.e., transformation codings of the
message body, such as gzip compression, that are acceptable in the
response.A server tests whether a content-coding is acceptable, according to
an Accept-Encoding field, using these rules:If the content-coding is one of the content-codings listed in
the Accept-Encoding field, then it is acceptable, unless it is
accompanied by a qvalue of 0. (As defined in Section 5.3.1 of
, a qvalue of 0 means "not
acceptable.")The special "*" symbol in an Accept-Encoding field matches any
available content-coding not explicitly listed in the header
field.If multiple content-codings are acceptable, then the acceptable
content-coding with the highest non-zero qvalue is preferred.The "identity" content-coding is always acceptable, i.e., no
transformation at all, unless specifically refused because the
Accept-Encoding field includes "identity;q=0" or because the field
includes "*;q=0" and does not explicitly include the "identity"
content-coding. If the Accept-Encoding field-value is empty, then
only the "identity" encoding is acceptable.If an Accept-Encoding field is present in a request, and if
the server cannot send a response that is acceptable according to the
Accept-Encoding header, then the server SHOULD send an error response
with the 406 (Not Acceptable) status code.If no Accept-Encoding field is present in a request, the server MAY
assume that the client will accept any content-coding. In this case,
if "identity" is one of the available content-codings, then the server
SHOULD use the "identity" content-coding, unless it has additional
information that a different content-coding is meaningful to the
client.The Accept-Language request-header field is similar to Accept, but
restricts the set of natural languages that are preferred as a
response to the request. Note that the language specified applies to
the presentation description (response message body) and any reason
phrases, but not the media content.A language tag identifies a natural language spoken, written, or
otherwise conveyed by human beings for communication of information to
other human beings. Computer languages are explicitly excluded. The
syntax and registry of RTSP 2.0 language tags are the same as those
defined by .Each language-range MAY be given an associated quality value that
represents an estimate of the user's preference for the languages
specified by that range. The quality value defaults to "q=1". For
example:Accept-Language: da, en-gb;q=0.8, en;q=0.7would mean: "I prefer Danish, but will accept British English and
other types of English." A language-range matches a language tag if it
exactly equals the full tag or if it exactly equals a prefix of the
tag, i.e., the primary-tag in the ABNF, such that the character
following primary-tag is "-". The special range "*", if present in the
Accept-Language field, matches every tag not matched by any other
range present in the Accept-Language field.Note: This use of a prefix matching rule does not imply that
language tags are assigned to languages in such a way that it is
always true that if a user understands a language with a certain
tag, then this user will also understand all languages with tags
for which this tag is a prefix. The prefix rule simply allows the
use of prefix tags if this is the case.In the process of selecting a language, each language tag is
assigned a qualification factor, i.e., if a language being supported
by the client is actually supported by the server and what
"preference" level the language achieves. The quality value (q-value)
of the longest language-range in the field that matches the language
tag is assigned as the qualification factor for a particular language
tag. If no language-range in the field matches the tag, the language
qualification factor assigned is 0. If no Accept-Language header is
present in the request, the server SHOULD assume that all languages
are equally acceptable. If an Accept-Language header is present, then
all languages that are assigned a qualification factor greater than 0
are acceptable.The Accept-Ranges general-header field allows indication of the
format supported in the Range header. The client MUST include the
header in SETUP requests to indicate which formats are acceptable when
received in PLAY and PAUSE responses and REDIRECT requests. The server
MUST include the header in SETUP responses and 456 (Header Field Not
Valid for Resource) error responses to indicate the formats supported
for the resource indicated by the Request-URI. The header MAY be
included in GET_PARAMETER request and response pairs. The
GET_PARAMETER request MUST contain a Session header to identify the
session context the request is related to. The requester and responder
will indicate their capabilities regarding Range formats
respectively.The syntax is defined in .The Allow message body header field lists the methods supported by
the resource identified by the Request-URI. The purpose of this field
is to inform the recipient of the complete set of valid methods
associated with the resource. An Allow header field MUST be present in
a 405 (Method Not Allowed) response. The Allow header MUST also be
present in all OPTIONS responses where the content of the header will
not include exactly the same methods as listed in the Public
header.The Allow message body header MUST also be included in SETUP and
DESCRIBE responses, if the methods allowed for the resource are
different from the complete set of methods defined in this memo.Example of use:The Authentication-Info response-header is used by the server to
communicate some information regarding the successful HTTP authentication in the response message.
The definition of the header is in , and any
applicable HTTP authentication schemes appear in other RFCs, such as
Digest. This header MUST only be used in
response messages related to client to server requests.An RTSP client that wishes to authenticate itself with a server
using the authentication mechanism from
HTTP, usually (but not necessarily) after receiving a 401
response, does so by including an Authorization request-header field
with the request. The Authorization field-value consists of
credentials containing the authentication information of the user
agent for the realm of the resource being requested. The definition of
the header is in , and any applicable HTTP
authentication schemes appear in other RFCs, such as Digest and Basic. This header MUST only be used in
client-to-server requests.If a request is authenticated and a realm specified, the same
credentials SHOULD be valid for all other requests within this realm
(assuming that the authentication scheme itself does not require
otherwise, such as credentials that vary according to a challenge
value or using synchronized clocks). Each client-to-server request
MUST be individually authorized by including the Authorization header
with the information.When a shared cache (see ) receives a
request containing an Authorization field, it MUST NOT return the
corresponding response as a reply to any other request, unless one of
the following specific exceptions holds:If the response includes the "max-age" cache directive, the
cache MAY use that response in replying to a subsequent request.
But (if the specified maximum age has passed) a proxy cache MUST
first revalidate it with the origin server, using the
request-headers from the new request to allow the origin server to
authenticate the new request. (This is the defined behavior for
max-age.) If the response includes "max-age=0", the proxy MUST
always revalidate it before reusing it.If the response includes the "must-revalidate" cache-control
directive, the cache MAY use that response in replying to a
subsequent request. But if the response is stale, all caches MUST
first revalidate it with the origin server, using the
request-headers from the new request to allow the origin server to
authenticate the new request.If the response includes the "public" cache directive, it MAY
be returned in reply to any subsequent request.The Bandwidth request-header field describes the estimated
bandwidth available to the client, expressed as a positive integer and
measured in kilobits per second. The bandwidth available to the client
may change during an RTSP session, e.g., due to mobility, congestion,
etc.Clients may not be able to accurately determine the available
bandwidth, for example, because the first hop is not a bottleneck.
Such a case is when the local area network (LAN) is not the
bottleneck, instead the LAN's Internet access link is, if the server
is not in the same LAN. Thus, link speeds of WLAN or Ethernet networks
are normally not a basis for estimating the available bandwidth.
Cellular devices or other devices directly connected to a modem or
connection-enabling device may more accurately estimate the bottleneck
bandwidth and what is a reasonable share of it for RTSP-controlled
media. The client will also need to take into account other traffic
sharing the bottleneck. For example, by only assigning a certain
fraction to RTSP and its media streams. It is RECOMMENDED that only
clients that have accurate and explicit information about bandwidth
bottlenecks use this header.This header is not a substitute for proper congestion control. It
is only a method providing an initial estimate and coarsely determines
if the selected content can be delivered at all.Example:The Blocksize request-header field is sent from the client to the
media server asking the server for a particular media packet size.
This packet size does not include lower-layer headers such as IP, UDP,
or RTP. The server is free to use a blocksize that is lower than the
one requested. The server MAY truncate this packet size to the closest
multiple of the minimum, media-specific block size or override it with
the media-specific size, if necessary. The block size MUST be a
positive decimal number measured in octets. The server only returns an
error (4xx) if the value is syntactically invalid.The Cache-Control general-header field is used to specify
directives that MUST be obeyed by all caching mechanisms along the
request/response chain.Cache directives MUST be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a
cache-directive for a specific cache.Cache-Control should only be specified in a DESCRIBE,
GET_PARAMETER, SET_PARAMETER, and SETUP request and its response.
Note: Cache-Control does not govern only the caching of responses for
the RTSP messages, instead it also applies to the media stream
identified by the SETUP request. The RTSP requests are generally not
cacheable; for further information, see .
Below are the descriptions of the cache directives that can be
included in the Cache-Control header.Indicates that the media stream or RTSP
response MUST NOT be cached anywhere. This allows an origin server
to prevent caching even by caches that have been configured to
return stale responses to client requests. Note: there is no
security function preventing the caching of content.Indicates that the media stream or RTSP
response is cacheable by any cache.Indicates that the media stream or RTSP
response is intended for a single user and MUST NOT be cached by a
shared cache. A private (non-shared) cache may cache the media
streams.An intermediate cache (proxy) may find
it useful to convert the media type of a certain stream. A proxy
might, for example, convert between video formats to save cache
space or to reduce the amount of traffic on a slow link. Serious
operational problems may occur, however, when these
transformations have been applied to streams intended for certain
kinds of applications. For example, applications for medical
imaging, scientific data analysis and those using end-to-end
authentication all depend on receiving a stream that is
bit-for-bit identical to the original media stream or RTSP
response. Therefore, if a response includes the no-transform
directive, an intermediate cache or proxy MUST NOT change the
encoding of the stream or response. Unlike HTTP, RTSP does not
provide for partial transformation at this point, e.g., allowing
translation into a different language.In some cases, such as times of
extremely poor network connectivity, a client may want a cache to
return only those media streams or RTSP responses that it
currently has stored and not to receive these from the origin
server. To do this, the client may include the only-if-cached
directive in a request. If the cache receives this directive, it
SHOULD either respond using a cached media stream or response that
is consistent with the other constraints of the request or respond
with a 504 (Gateway Timeout) status. However, if a group of caches
is being operated as a unified system with good internal
connectivity, such a request MAY be forwarded within that group of
caches.Indicates that the client is willing to
accept a media stream or RTSP response that has exceeded its
expiration time. If max-stale is assigned a value, then the client
is willing to accept a response that has exceeded its expiration
time by no more than the specified number of seconds. If no value
is assigned to max-stale, then the client is willing to accept a
stale response of any age.Indicates that the client is willing to
accept a media stream or RTSP response whose freshness lifetime is
no less than its current age plus the specified time in seconds.
That is, the client wants a response that will still be fresh for
at least the specified number of seconds.When the must-revalidate directive
is present in a SETUP response received by a cache, that cache
MUST NOT use the cache entry after it becomes stale to respond to
a subsequent request without first revalidating it with the origin
server. That is, the cache is required to do an end-to-end
revalidation every time, if, based solely on the origin server's
Expires, the cached response is stale.The proxy-revalidate directive has
the same meaning as the must-revalidate directive, except that it
does not apply to non-shared user agent caches. It can be used on
a response to an authenticated request to permit the user's cache
to store and later return the response without needing to
revalidate it (since it has already been authenticated once by
that user), while still requiring proxies that service many users
to revalidate each time (in order to make sure that each user has
been authenticated). Note that such authenticated responses also
need the "public" cache directive in order to allow them to be
cached at all.When an intermediate cache is forced, by
means of a max-age=0 directive, to revalidate its own cache entry,
and the client has supplied its own validator in the request, the
supplied validator might differ from the validator currently
stored with the cache entry. In this case, the cache MAY use
either validator in making its own request without affecting
semantic transparency.However, the choice of validator might affect
performance. The best approach is for the intermediate cache to
use its own validator when making its request. If the server
replies with 304 (Not Modified), then the cache can return its now
validated copy to the client with a 200 (OK) response. If the
server replies with a new message body and cache validator,
however, the intermediate cache can compare the returned validator
with the one provided in the client's request, using the strong
comparison function. If the client's validator is equal to the
origin server's, then the intermediate cache simply returns 304
(Not Modified). Otherwise, it returns the new message body with a
200 (OK) response.The Connection general-header field allows the sender to specify
options that are desired for that particular connection. It MUST NOT
be communicated by proxies over further connections.RTSP 2.0 proxies MUST parse the Connection header field before a
message is forwarded and, for each connection-token in this field,
remove any header field(s) from the message with the same name as the
connection-token. Connection options are signaled by the presence of a
connection-token in the Connection header field, not by any
corresponding additional header field(s), since the additional header
field may not be sent if there are no parameters associated with that
connection option.Message headers listed in the Connection header MUST NOT include
end-to-end headers, such as Cache-Control.RTSP 2.0 defines the "close" connection option for the sender to
signal that the connection will be closed after completion of the
response. For example, "Connection: close in either the request or the
response-header fields" indicates that the connection SHOULD NOT be
considered "persistent"
after the current request/response is complete.The use of the connection option "close" in RTSP messages SHOULD be
limited to error messages when the server is unable to recover and
therefore sees it necessary to close the connection. The reason being
that the client has the choice of continuing using a connection
indefinitely, as long as it sends valid messages.The Connection-Credentials response-header is used to carry the
chain of credentials for any next hop that needs to be approved by the
requester. It MUST only be used in server-to-client responses.The Connection-Credentials header in an RTSP response MUST, if
included, contain the credential information (in the form of a list of
certificates providing the chain of certification) of the next hop to
which an intermediary needs to securely connect. The header MUST
include the URI of the next hop (proxy or server) and a Base64-encoded
(according to Section 4 of and where the
padding bits are set to zero) binary structure containing a sequence
of DER-encoded X.509v3 certificates .The binary structure starts with the number of certificates
(NR_CERTS) included as a 16-bit unsigned integer. This is followed by
an NR_CERTS number of 16-bit unsigned integers providing the size, in
octets, of each DER-encoded certificate. This is followed by an
NR_CERTS number of DER-encoded X.509v3 certificates in a sequence
(chain). This format is exemplified in . The certificate of the proxy or
server must come first in the structure. Each following certificate
must directly certify the one preceding it. Because certificate
validation requires that root keys be distributed independently, the
self-signed certificate that specifies the root certificate authority
may optionally be omitted from the chain, under the assumption that
the remote end must already possess it in order to validate it in any
case.Example:Where MIIDNTCC... is a Base64 encoding of the following
structure:The Content-Base message body header field may be used to specify
the base URI for resolving relative URIs within the message body.If no Content-Base field is present, the base URI of a message body
is defined by either its Content-Location (if that Content-Location
URI is an absolute URI) or the URI used to initiate the request, in
that order of precedence. Note, however, that the base URI of the
contents within the message body may be redefined within that message
body.The Content-Encoding message body header field is used as a
modifier of the media-type. When present, its value indicates what
additional content-codings have been applied to the message body, and
thus what decoding mechanisms must be applied in order to obtain the
media-type referenced by the Content-Type header field.
Content-Encoding is primarily used to allow a document to be
compressed without losing the identity of its underlying media
type.The content-coding is a characteristic of the message body
identified by the Request-URI. Typically, the message body is stored
with this encoding and is only decoded before rendering or analogous
usage. However, an RTSP proxy MAY modify the content-coding if the new
coding is known to be acceptable to the recipient, unless the
"no-transform" cache directive is present in the message.If the content-coding of a message body is not "identity", then the
message MUST include a Content-Encoding message body header that lists
the non-identity content-coding(s) used.If the content-coding of a message body in a request message is not
acceptable to the origin server, the server SHOULD respond with a
status code of 415 (Unsupported Media Type).If multiple encodings have been applied to a message body, the
content-codings MUST be listed in the order in which they were
applied, first to last from left to right. Additional information
about the encoding parameters MAY be provided by other header fields
not defined by this specification.The Content-Language message body header field describes the
natural language(s) of the intended audience for the enclosed message
body. Note that this might not be equivalent to all the languages used
within the message body.Language tags are mentioned in . The primary purpose of
Content-Language is to allow a user to identify and differentiate
entities according to the user's own preferred language. Thus, if the
body content is intended only for a Danish-literate audience, the
appropriate field isContent-Language: daIf no Content-Language is specified, the default is that the
content is intended for all language audiences. This might mean that
the sender does not consider it to be specific to any natural language
or that the sender does not know for which language it is
intended.Multiple languages MAY be listed for content that is intended for
multiple audiences. For example, a rendition of the "Treaty of
Waitangi", presented simultaneously in the original Maori and English
versions, would call forContent-Language: mi, enHowever, just because multiple languages are present within a
message body does not mean that it is intended for multiple linguistic
audiences. An example would be a beginner's language primer, such as
"A First Lesson in Latin", which is clearly intended to be used by an
English-literate audience. In this case, the Content-Language would
properly only include "en".Content-Language MAY be applied to any media type -- it is not
limited to textual documents.The Content-Length message body header field contains the length of
the message body of the RTSP message (i.e., after the double CRLF
following the last header) in octets of bits. Unlike HTTP, it MUST be
included in all messages that carry a message body beyond the header
portion of the RTSP message. If it is missing, a default value of zero
is assumed. Any Content-Length greater than or equal to zero is a
valid value.The Content-Location message body header field MAY be used to
supply the resource location for the message body enclosed in the
message when that body is accessible from a location separate from the
requested resource's URI. A server SHOULD provide a Content-Location
for the variant corresponding to the response message body; especially
in the case where a resource has multiple variants associated with it,
and those entities actually have separate locations by which they
might be individually accessed, the server SHOULD provide a
Content-Location for the particular variant that is returned.As an example, if an RTSP client performs a DESCRIBE request on a
given resource, e.g.,
"rtsp://a.example.com/movie/Plan9FromOuterSpace", then the server may
use additional information, such as the User-Agent header, to
determine the capabilities of the agent. The server will then return a
media description tailored to that class of RTSP agents. To indicate
which specific description the agent receives, the resource identifier
("rtsp://a.example.com/movie/Plan9FromOuterSpace/FullHD.sdp") is
provided in Content-Location, while the description is still a valid
response for the generic resource identifier, thus enabling both
debugging and cache operation as discussed below.The Content-Location value is not a replacement for the original
requested URI; it is only a statement of the location of the resource
corresponding to this particular variant at the time of the request.
Future requests MAY specify the Content-Location URI as the
Request-URI if the desire is to identify the source of that particular
variant. This is useful if the RTSP agent desires to verify if the
resource variant is current through a conditional request.A cache cannot assume that a message body with a Content-Location
different from the URI used to retrieve it can be used to respond to
later requests on that Content-Location URI. However, the
Content-Location can be used to differentiate between multiple
variants retrieved from a single requested resource.If the Content-Location is a relative URI, the relative URI is
interpreted relative to the Request-URI.Note that Content-Location can be used in some cases to derive the
base-URI for relative URI(s) present in session description formats.
This needs to be taken into account when Content-Location is used. The
easiest way to avoid needing to consider that issue is to include the
Content-Base whenever the Content-Location is included.Note also, when using Media Tags in conjunction with
Content-Location, it is important that the different versions have
different MTags, even if provided under different Content-Location
URIs. This is because the different content variants still have been
provided in response to the same request URI.Note also, as in most cases, the URIs used in the DESCRIBE and the
SETUP requests are different: the URI provided in a DESCRIBE
Content-Location response can't directly be used in a SETUP request.
Instead, the steps of deriving the media resource URIs are necessary.
This commonly involves combing the media description's relative URIs,
e.g., from the SDP's a=control attribute, with the base-URI to create
the absolute URIs needed in the SETUP request.The Content-Type message body header indicates the media type of
the message body sent to the recipient. Note that the content types
suitable for RTSP are likely to be restricted in practice to
presentation descriptions and parameter-value types.The CSeq general-header field specifies the sequence number
(integer) for an RTSP request/response pair. This field MUST be
present in all requests and responses. RTSP agents maintain a sequence
number series for each responder to which they have an open message
transport channel. For each new RTSP request an agent originates on a
particular RTSP message transport, the CSeq value MUST be incremented
by one. The initial sequence number can be any number; however, it is
RECOMMENDED to start at 0. Each sequence number series is unique
between each requester and responder, i.e., the client has one series
for its requests to a server and the server has another when sending
requests to the client. Each requester and responder is identified by
its socket address (IP address and port number), i.e., per direction
of a TCP connection. Any retransmitted request MUST contain the same
sequence number as the original, i.e., the sequence number is not
incremented for retransmissions of the same request. The RTSP agent
receiving requests MUST process the requests arriving on a particular
transport in the order of the sequence numbers. Responses are sent in
the order that they are generated. The RTSP response MUST have the
same sequence number as was present in the corresponding request. An
RTSP agent receiving a response MAY receive the responses out of order
compared to the order of the requests it sent. Thus, the agent MUST
use the sequence number in the response to pair it with the
corresponding request.The main purpose of the sequence number is to map responses to
requests.The requirement to use a sequence-number increment of one for
each new request is to support any future specification of RTSP
message transport over a protocol that does not provide in-order
delivery or is unreliable.The above rules relating to the initial sequence number may
appear unnecessarily loose. The reason for this is to cater to
some common behavior of existing implementations: when using
multiple reliable connections in sequence, it may still be easiest
to use a single sequence-number series for a client connecting
with a particular server. Thus, the initial sequence number may be
arbitrary depending on the number of previous requests. For any
unreliable transport, a stricter definition or other solution will
be required to enable detection of any loss of the first
request.When using multiple sequential transport connections, there is
no protocol mechanism to ensure in-order processing as the
sequence number is scoped on the individual transport connection
and its five tuple. Thus, there are potential issues with opening
a new transport connection to the same host for which there
already exists a transport connection with outstanding requests
and previously dispatched requests related to the same RTSP
session.RTSP Proxies also need to follow the above rules. This implies that
proxies that aggregate requests from multiple clients onto a single
transport towards a server or a next-hop proxy need to renumber these
requests to form a unified sequence on that transport, fulfilling the
above rules. A proxy capable of fulfilling some agent's request
without emitting its own request (e.g., a caching proxy that fulfills
a request from its cache) also causes a need to renumber as the number
of received requests with a particular target may not be the same as
the number of emitted requests towards that target agent. A proxy that
needs to renumber needs to perform the corresponding renumbering back
to the original sequence number for any received response before
forwarding it back to the originator of the request.A client connected to a proxy, and using that transport to send
requests to multiple servers, creates a situation where it is
quite likely to receive the responses out of order. This is
because the proxy will establish separate transports from the
proxy to the servers on which to forward the client's requests.
When the responses arrive from the different servers, they will be
forwarded to the client in the order they arrive at the proxy and
can be processed, not the order of the client's original sequence
numbers. This is intentional to avoid some session's requests
being blocked by another server's slow processing of requests.The Date general-header field represents the date and time at which
the message was originated. The inclusion of the Date header in an
RTSP message follows these rules:An RTSP message, sent by either the client or the server,
containing a body MUST include a Date header, if the sending host
has a clock;Clients and servers are RECOMMENDED to include a Date header in
all other RTSP messages, if the sending host has a clock;If the server does not have a clock that can provide a
reasonable approximation of the current time, its responses MUST
NOT include a Date header field. In this case, this rule MUST be
followed: some origin-server implementations might not have a
clock available. An origin server without a clock MUST NOT assign
Expires or Last-Modified values to a response, unless these values
were associated with the resource by a system or user with a
reliable clock. It MAY assign an Expires value that is known, at
or before server-configuration time, to be in the past (this
allows "pre-expiration" of responses without storing separate
Expires values for each resource).A received message that does not have a Date header field MUST be
assigned one by the recipient if the message will be cached by that
recipient. An RTSP implementation without a clock MUST NOT cache
responses without revalidating them on every use. An RTSP cache,
especially a shared cache, SHOULD use a mechanism, such as the Network Time Protocol (NTP), to
synchronize its clock with a reliable external standard.The RTSP-date, a full date as specified by Section 3.3 of , sent in a Date header SHOULD NOT represent a date
and time subsequent to the generation of the message. It SHOULD
represent the best available approximation of the date and time of
message generation, unless the implementation has no means of
generating a reasonably accurate date and time. In theory, the date
ought to represent the moment just before the message body is
generated. In practice, the date can be generated at any time during
the message origination without affecting its semantic value.Note: The RTSP 2.0 date format is defined to be the full-date
format in RFC 5322. This format is more flexible than the date
format in RFC 1123 used by RTSP 1.0. Thus, implementations should
use single spaces as separators, as recommended by RFC 5322, and
support receiving the obsolete format.Further, note that the syntax allows for a comment to be added
at the end of the date.The Expires message body header field gives a date and time after
which the description or media-stream should be considered stale. The
interpretation depends on the method: The Expires header indicates a
date and time after which the presentation description (body)
SHOULD be considered stale.The Expires header indicates a date
and time after which the media stream SHOULD be considered
stale.A stale cache entry should not be returned by a cache (either a
proxy cache or a user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh copy
of the message body). See for further
discussion of the expiration model.The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.The format is an absolute date and time as defined by RTSP-date. An
example of its use isRTSP 2.0 clients and caches MUST treat other invalid date formats,
especially those including the value "0", as having occurred in the
past (i.e., already expired).To mark a response as "already expired," an origin server should
use an Expires date that is equal to the Date header value. To mark a
response as "never expires", an origin server SHOULD use an Expires
date approximately one year from the time the response is sent. RTSP
2.0 servers SHOULD NOT send Expires dates that are more than one year
in the future.The From request-header field, if given, SHOULD contain an Internet
email address for the human user who controls the requesting user
agent. The address SHOULD be machine usable, as defined by "mailbox"
in .This header field MAY be used for logging purposes and as a means
for identifying the source of invalid or unwanted requests. It SHOULD
NOT be used as an insecure form of access protection. The
interpretation of this field is that the request is being performed on
behalf of the person given, who accepts responsibility for the method
performed. In particular, robot agents SHOULD include this header so
that the person responsible for running the robot can be contacted if
problems occur on the receiving end.The Internet email address in this field MAY be separate from the
Internet host that issued the request. For example, when a request is
passed through a proxy, the original issuer's address SHOULD be
used.The client SHOULD NOT send the From header field without the user's
approval, as it might conflict with the user's privacy interests or
their site's security policy. It is strongly recommended that the user
be able to disable, enable, and modify the value of this field at any
time prior to a request.The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, independent of how the
presentation description was received. The presentation description
can be fetched via means external to RTSP (such as HTTP) or via the
DESCRIBE message. In the case of retrieving the presentation
description via RTSP, the server implementation is guaranteeing the
integrity of the description between the time of the DESCRIBE message
and the SETUP message. By including the MTag given in or with the
session description in an If-Match header part of the SETUP request,
the client ensures that resources set up are matching the description.
A SETUP request with the If-Match header for which the MTag validation
check fails MUST generate a response using 412 (Precondition
Failed).This validation check is also very useful if a session has been
redirected from one server to another.The If-Modified-Since request-header field is used with the
DESCRIBE and SETUP methods to make them conditional. If the requested
variant has not been modified since the time specified in this field,
a description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response MUST be returned without any message body.An example of the field is:This request-header can be used with one or several message body
tags to make DESCRIBE requests conditional. A client that has one or
more message bodies previously obtained from the resource can verify
that none of those entities is current by including a list of their
associated message body tags in the If-None-Match header field. The
purpose of this feature is to allow efficient updates of cached
information with a minimum amount of transaction overhead. As a
special case, the value "*" matches any current entity of the
resource.If any of the message body tags match the message body tag of the
message body that would have been returned in the response to a
similar DESCRIBE request (without the If-None-Match header) on that
resource, or if "*" is given and any current entity exists for that
resource, then the server MUST NOT perform the requested method,
unless required to do so because the resource's modification date
fails to match that supplied in an If-Modified-Since header field in
the request. Instead, if the request method was DESCRIBE, the server
SHOULD respond with a 304 (Not Modified) response, including the
cache-related header fields (particularly MTag) of one of the message
bodies that matched. For all other request methods, the server MUST
respond with a status of 412 (Precondition Failed).See for rules on how to
determine if two message body tags match.If none of the message body tags match, then the server MAY perform
the requested method as if the If-None-Match header field did not
exist, but MUST also ignore any If-Modified-Since header field(s) in
the request. That is, if no message body tags match, then the server
MUST NOT return a 304 (Not Modified) response.If the request would, without the If-None-Match header field,
result in anything other than a 2xx or 304 status, then the
If-None-Match header MUST be ignored. (See for a discussion of server behavior
when both If-Modified-Since and If-None-Match appear in the same
request.)The result of a request having both an If-None-Match header field
and an If-Match header field is unspecified and MUST be considered an
illegal request.The Last-Modified message body header field indicates the date and
time at which the origin server believes the presentation description
or media stream was last modified. For the DESCRIBE method, the header
field indicates the last modification date and time of the
description, for the SETUP of the media stream.An origin server MUST NOT send a Last-Modified date that is later
than the server's time of message origination. In such cases, where
the resource's last modification would indicate some time in the
future, the server MUST replace that date with the message origination
date.An origin server SHOULD obtain the Last-Modified value of the
message body as close as possible to the time that it generates the
Date value of its response. This allows a recipient to make an
accurate assessment of the message body's modification time,
especially if the message body changes near the time that the response
is generated.RTSP servers SHOULD send Last-Modified whenever feasible.The Location response-header field is used to redirect the
recipient to a location other than the Request-URI for completion of
the request or identification of a new resource. For 3rr responses,
the location SHOULD indicate the server's preferred URI for automatic
redirection to the resource. The field-value consists of a single
absolute URI.Note: The Content-Location
header field differs from Location in that the Content-Location
identifies the original location of the message body enclosed in the
request. Therefore, it is possible for a response to contain header
fields for both Location and Content-Location. Also, see for cache requirements of some
methods.This general-header is used in SETUP responses or PLAY_NOTIFY
requests to indicate the media's properties that currently are
applicable to the RTSP session. PLAY_NOTIFY MAY be used to modify
these properties at any point. However, the client SHOULD have
received the update prior to any action related to the new media
properties taking effect. For aggregated sessions, the
Media-Properties header will be returned in each SETUP response. The
header received in the latest response is the one that applies on the
whole session from this point until any future update. The header MAY
be included without value in GET_PARAMETER requests to the server with
a Session header included to query the current Media-Properties for
the session. The responder MUST include the current session's media
properties.The media properties expressed by this header are the ones
applicable to all media in the RTSP session. For aggregated sessions,
the header expressed the combined media-properties. As a result,
aggregation of media MAY result in a change of the media properties
and, thus, the content of the Media-Properties header contained in
subsequent SETUP responses.The header contains a list of property values that are applicable
to the currently setup media or aggregate of media as indicated by the
RTSP URI in the request. No ordering is enforced within the header.
Property values should be placed into a single group that handles a
particular orthogonal property. Values or groups that express multiple
properties SHOULD NOT be used. The list of properties that can be
expressed MAY be extended at any time. Unknown property values MUST be
ignored.This specification defines the following four groups and their
property values:Indicates that random access is
possible. May optionally include a floating-point value in
seconds indicating the longest duration between any two random
access points in the media.Seeking is limited to the
beginning only.No seeking is possible.The content will not be changed
during the lifetime of the RTSP session.The content may be changed based on
external methods or triggers.The media accessible
progresses as wallclock time progresses.Content will be retained for the
duration of the lifetime of the RTSP session.Content will be retained at least
until the specified wallclock time. The time must be provided
in the absolute time format specified in .Each individual media unit is
retained for at least the specified Time-Duration. This
definition allows for retaining data with a time-based sliding
window. The time duration is expressed as floating-point
number in seconds. The value 0.0 is a valid as this indicates
that no data is retained in a time-progressing session.A quoted comma-separated list of one or
more decimal values or ranges of scale values supported by the
content in arbitrary order. A range has a start and stop value
separated by a colon. A range indicates that the content
supports a fine-grained selection of scale values.
Fine-graining allows for steps at least as small as one tenth
of a scale value. Content is considered to support
fine-grained selection when the server in response to a given
scale value can produce content with an actual scale that is
less than one tenth of scale unit, i.e., 0.1, from the
requested value. Negative values are supported. The value 0
has no meaning and MUST NOT be used.Examples of this header for on-demand content and a live stream
without recording are:The Media-Range general-header is used to give the range of the
media at the time of sending the RTSP message. This header MUST be
included in the SETUP response, PLAY and PAUSE responses for media
that are time-progressing, PLAY and PAUSE responses after any change
for media that are Dynamic, and in PLAY_NOTIFY requests that are sent
due to Media-Property-Update. A Media-Range header without any range
specifications MAY be included in GET_PARAMETER requests to the server
to request the current range. In this case, the server MUST include
the current range at the time of sending the response.The header MUST include range specifications for all time formats
supported for the media, as indicated in Accept-Ranges header when setting up
the media. The server MAY include more than one range specification of
any given time format to indicate media that has non-continuous range.
The range specifications SHALL be ordered with the range with the
lowest value or earliest start time first, followed by ranges with
increasingly higher values or later start time.For media that has the time-progressing property, the Media-Range
header values will only be valid for the particular point in time when
it was issued. As the wallclock progresses, so will the media range.
However, it shall be assumed that media time progresses in direct
relationship to wallclock time (with the exception of clock skew) so
that a reasonably accurate estimation of the media range can be
calculated.The MTag response-header MAY be included in DESCRIBE,
GET_PARAMETER, or SETUP responses. The message body tags () returned in a DESCRIBE response and the
one in SETUP refer to the presentation, i.e., both the returned
session description and the media stream. This allows for verification
that one has the right session description to a media resource at the
time of the SETUP request. However, it has the disadvantage that a
change in any of the parts results in invalidation of all the
parts.If the MTag is provided both inside the message body, e.g., within
the "a=mtag" attribute in SDP, and in the response message, then both
tags MUST be identical. It is RECOMMENDED that the MTag be primarily
given in the RTSP response message, to ensure that caches can use the
MTag without requiring content inspection. However, for session
descriptions that are distributed outside of RTSP, for example, using
HTTP, etc., it will be necessary to include the message body tag in
the session description as specified in .SETUP and DESCRIBE requests can be made conditional upon the MTag
using the headers If-Match () and
If-None-Match ().The Notify-Reason response-header is solely used in the PLAY_NOTIFY
method. It indicates the reason why the server has sent the
asynchronous PLAY_NOTIFY request (see ).The Pipelined-Requests general-header is used to indicate that a
request is to be executed in the context created by a previous
request(s). The primary usage of this header is to allow pipelining of
SETUP requests so that any additional SETUP request after the first
one does not need to wait for the session ID to be sent back to the
requesting agent. The header contains a unique identifier that is
scoped by the persistent connection used to send the requests.Upon receiving a request with the Pipelined-Requests, the
responding agent MUST look up if there exists a binding between this
Pipelined-Requests identifier for the current persistent connection
and an RTSP session ID. If the binding exists, then the received
request is processed the same way as if it contained the Session
header with the found session ID. If there does not exist a mapping
and no Session header is included in the request, the responding agent
MUST create a binding upon the successful completion of a session
creating request, i.e., SETUP. A binding MUST NOT be created, if the
request failed to create an RTSP session. In case the request contains
both a Session header and the Pipelined-Requests header, the
Pipelined-Requests header MUST be ignored.Note: Based on the above definition, at least the first request
containing a new unique Pipelined-Requests header will be required to
be a SETUP request (unless the protocol is extended with new methods
of creating a session). After that first one, additional SETUP
requests or requests of any type using the RTSP session context may
include the Pipelined-Requests header.When responding to any request that contained the
Pipelined-Requests header, the server MUST also include the Session
header when a binding to a session context exists. An RTSP agent that
knows the session identifier SHOULD NOT use the Pipelined-Requests
header in any request and only use the Session header. This as the
Session identifier is persistent across transport contexts, like TCP
connections, which the Pipelined-Requests identifier is not.The RTSP agent sending the request with a Pipelined-Requests header
has the responsibility for using a unique and previously unused
identifier within the transport context. Currently, only a TCP
connection is defined as such a transport context. A server MUST
delete the Pipelined-Requests identifier and its binding to a session
upon the termination of that session. Despite the previous mandate,
RTSP agents are RECOMMENDED not to reuse identifiers to allow for
better error handling and logging.RTSP Proxies may need to translate Pipelined-Requests identifier
values from incoming requests to outgoing to allow for aggregation of
requests onto a persistent connection.The Proxy-Authenticate response-header field MUST be included as
part of a 407 (Proxy Authentication Required) response. The
field-value consists of a challenge that indicates the authentication
scheme and parameters applicable to the proxy for this Request-URI.
The definition of the header is in , and any
applicable HTTP authentication schemes appear in other RFCs, such as
Digest and Basic.The HTTP access authentication process is described in . This header MUST only be used in response messages
related to client-to-server requests.The Proxy-Authentication-Info response-header is used by the proxy
to communicate some information regarding the successful
authentication to the proxy in the message response in some
authentication schemes, such as the Digest
scheme. The definition of the header is in , and any applicable HTTP authentication schemes
appear in other RFCs. This header MUST only be used in response
messages related to client-to-server requests. This header has
hop-by-hop scope.The Proxy-Authorization request-header field allows the client to
identify itself (or its user) to a proxy that requires authentication.
The Proxy-Authorization field-value consists of credentials containing
the authentication information of the user agent for the proxy or
realm of the resource being requested. The definition of the header is
in , and any applicable HTTP authentication
schemes appear in other RFCs, such as Digest and Basic.The HTTP access authentication process is described in . Unlike Authorization, the Proxy-Authorization
header field applies only to the next-hop proxy. This header MUST only
be used in client-to-server requests.The Proxy-Require request-header field is used to indicate
proxy-sensitive features that MUST be supported by the proxy. Any
Proxy-Require header features that are not supported by the proxy MUST
be negatively acknowledged by the proxy to the client using the
Unsupported header. The proxy MUST use the 551 (Option Not Supported)
status code in the response. Any feature tag included in the
Proxy-Require does not apply to the endpoint (server or client). To
ensure that a feature is supported by both proxies and servers, the
tag needs to be included in also a Require header.See for more details on the mechanics
of this message and a usage example. See discussion in the proxies section about when to consider
that a feature requires proxy support.Example of use:The Proxy-Supported general-header field enumerates all the
extensions supported by the proxy using feature tags. The header
carries the intersection of extensions supported by the forwarding
proxies. The Proxy-Supported header MAY be included in any request by
a proxy. It MUST be added by any proxy if the Supported header is
present in a request. When present in a request, the receiver MUST
copy the received Proxy-Supported header in the response.The Proxy-Supported header field contains a list of feature tags
applicable to proxies, as described in . The list is the intersection of all
feature tags understood by the proxies. To achieve an intersection,
the proxy adding the Proxy-Supported header includes all proxy feature
tags it understands. Any proxy receiving a request with the header
MUST check the list and remove any feature tag(s) it does not support.
A Proxy-Supported header present in the response MUST NOT be modified
by the proxies. These feature tags are the ones the proxy chains
support in general and are not specific to the request resource.Example:The Public response-header field lists the set of methods supported
by the response sender. This header applies to the general
capabilities of the sender, and its only purpose is to indicate the
sender's capabilities to the recipient. The methods listed may or may
not be applicable to the Request-URI; the Allow header field MAY be used to indicate
methods allowed for a particular URI.Example of use:In the event that there are proxies between the sender and the
recipient of a response, each intervening proxy MUST modify the Public
header field to remove any methods that are not supported via that
proxy. The resulting Public header field will contain an intersection
of the sender's methods and the methods allowed through by the
intervening proxies.In general, proxies should allow all methods to transparently
pass through from the sending RTSP agent to the receiving RTSP
agent, but there may be cases where this is not desirable for a
given proxy. Modification of the Public response-header field by
the intervening proxies ensures that the request sender gets an
accurate response indicating the methods that can be used on the
target agent via the proxy chain.The Range general-header specifies a time range in PLAY (), PAUSE (), SETUP (), and PLAY_NOTIFY () requests and responses. It MAY be included
in GET_PARAMETER requests from the client to the server with only a
Range format and no value to request the current media position,
whether the session is in Play or Ready state in the included format.
The server SHALL, if supporting the range format, respond with the
current playing point or pause point as the start of the range. If an
explicit stop point was used in the previous PLAY request, then that
value shall be included as stop point. Note that if the server is
currently under any type of media playback manipulation affecting the
interpretation of the Range header, like scale value other than 1,
that fact is also required to be included in any GET_PARAMETER
response by including the Scale header to provide complete
information.The range can be specified in a number of units. This specification
defines smpte (), npt (), and clock () range
units. While octet ranges (Byte Ranges) (see Section 2.1 of ) and other extended units MAY be used, their
behavior is unspecified since they are not normally meaningful in
RTSP. Servers supporting the Range header MUST understand the NPT
range format and SHOULD understand the SMPTE range format. If the
Range header is sent in a time format that is not understood, the
recipient SHOULD return 456 (Header Field Not Valid for Resource) and
include an Accept-Ranges header indicating the supported time formats
for the given resource.Example:The Range header contains a range of one single range format. A
range is a half-open interval with a start and an end point, including
the start point but excluding the end point. A range may either be
fully specified with explicit values for start point and end point or
have either the start or end point be implicit. An implicit start
point indicates the session's pause point, and if no pause point is
set, the start of the content. An implicit end point indicates the end
of the content. The usage of both implicit start and end points is not
allowed in the same Range header; however, the omission of the Range
header has that meaning, i.e., from pause point (or start) until end
of content.As noted, Range headers define half-open intervals. A range of
A-B starts exactly at time A, but ends just before B. Only the
start time of a media unit such as a video or audio frame is
relevant. For example, assume that video frames are generated
every 40 ms. A range of 10.0-10.1 would include a video frame
starting at 10.0 or later time and would include a video frame
starting at 10.08, even though it lasted beyond the interval. A
range of 10.0-10.08, on the other hand, would exclude the frame at
10.08.Please note the difference between NPT timescales' "now" and an
implicit start value. Implicit values reference the current
pause-point, while "now" is the current time. In a
time-progressing session with recording (retention for some or
full time), the pause point may be 2 min into the session while
now could be 1 hour into the session.By default, range intervals increase, where the second point is
larger than the first point.Example:However, range intervals can also decrease if the Scale header (see
) indicates a negative scale value. For
example, this would be the case when a playback in reverse is
desired.Example:Decreasing ranges are still half-open intervals as described above.
Thus, for range A-B, A is closed and B is open. In the above example,
15 is closed and 10 is open. An exception to this rule is the case
when B=0 is in a decreasing range. In this case, the range is closed
on both ends, as otherwise there would be no way to reach 0 on a
reverse playback for formats that have such a notion, like NPT and
SMPTE.Example:In this range, both 15 and 0 are closed.A decreasing range interval without a corresponding negative value
in the Scale header is not valid.The Referrer request-header field allows the client to specify, for
the server's benefit, the address (URI) of the resource from which the
Request-URI was obtained. The URI refers to that of the presentation
description, typically retrieved via HTTP. The Referrer request-header
allows a server to generate lists of back-links to resources for
interest, logging, optimized caching, etc. It also allows obsolete or
mistyped links to be traced for maintenance. The Referrer field MUST
NOT be sent if the Request-URI was obtained from a source that does
not have its own URI, such as input from the user keyboard.If the field-value is a relative URI, it SHOULD be interpreted
relative to the Request-URI. The URI MUST NOT include a fragment
identifier.Because the source of a link might be private information or might
reveal an otherwise private information source, it is strongly
recommended that the user be able to select whether or not the
Referrer field is sent. For example, a streaming client could have a
toggle switch for openly/anonymously, which would respectively
enable/disable the sending of Referrer and From information.Clients SHOULD NOT include a Referrer header field in an
(non-secure) RTSP request if the referring page was transferred with a
secure protocol.This request-header is used to indicate the end result for requests
that take time to complete, such as PLAY. It is sent in PLAY_NOTIFY with the end-of-stream
reason to report how the PLAY request concluded, either in success or
in failure. The header carries a reference to the request it reports
on using the CSeq number and the Session ID used in the request
reported on. This is not ensured to be unambigous due to the fact that
the CSeq number is scoped by the transport connection. Agents
originating requests can reduce the issue by using a monotonically
increasing counter across all sequential transports used. The header
provides both a numerical status code (according to ) and a human-readable reason phrase.Proxies that renumber the CSeq header need to perform corresponding
remapping of the cseq parameter in this header when forwarding the
request to the next-hop agent.The Require request-header field is used by agents to ensure that
the other endpoint supports features that are required in respect to
this request. It can also be used to query if the other endpoint
supports certain features; however, the use of the Supported
general-header () is much more effective
in this purpose. In case any of the feature tags listed by the Require
header are not supported by the server or client receiving the
request, it MUST respond to the request using the error code 551
(Option Not Supported) and include the Unsupported header listing
those feature tags that are NOT supported. This header does not apply
to proxies; for the same functionality with respect to proxies, see
the Proxy-Require header () with the
exception of media-modifying proxies. Media-modifying proxies, due to
their nature of handling media in a way that is very similar to a
server, do need to understand also the server's features to correctly
serve the client.This is to make sure that the client-server interaction will
proceed without delay when all features are understood by both
sides and only slow down if features are not understood (as in the
example below). For a well-matched client-server pair, the
interaction proceeds quickly, saving a round trip often required
by negotiation mechanisms. In addition, it also removes state
ambiguity when the client requires features that the server does
not understand.Example (Not complete):In this example, "funky-feature" is the feature tag that indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.Proxies and other intermediary devices MUST ignore this header. If
a particular extension requires that intermediate devices support it,
the extension should be tagged in the Proxy-Require field instead (see
). See discussion in the proxies section about when to consider
that a feature requires proxy support.The Retry-After response-header field can be used with a 503
(Service Unavailable) or 553 (Proxy Unavailable) response to indicate
how long the service is expected to be unavailable to the requesting
client. This field MAY also be used with any 3rr (Redirection)
response to indicate the minimum time the user agent is asked to wait
before issuing the redirected request. A response using 413 (Request
Message Body Too Large) when the restriction is temporary MAY also
include the Retry-After header. The value of this field can be either
an RTSP-date or an integer number of seconds (in decimal) after the
time of the response.Example:In the latter example, the delay is 2 minutes.The RTP-Info general-header field is used to set RTP-specific
parameters in the PLAY and GET_PARAMETER responses or PLAY_NOTIFY and
GET_PARAMETER requests. For streams using RTP as transport protocol,
the RTP-Info header SHOULD be part of a 200 response to PLAY.The exclusion of the RTP-Info in a PLAY response for
RTP-transported media will result in a client needing to
synchronize the media streams using RTCP. This may have negative
impact as the RTCP can be lost and does not need to be
particularly timely in its arrival. Also, functionality that
informs the client from which packet a seek has occurred is
affected.The RTP-Info MAY be included in SETUP responses to provide
synchronization information when changing transport parameters, see
. The RTP-Info header and the Range header
MAY be included in a GET_PARAMETER request from client to server
without any values to request the current playback point and
corresponding RTP synchronization information. When the RTP-Info
header is included in a Request, the Range header MUST also be
included. The server response SHALL include both the Range header and
the RTP-Info header. If the session is in Play state, then the value
of the Range header SHALL be filled in with the current playback point
and with the corresponding RTP-Info values. If the server is in
another state, no values are included in the RTP-Info header. The
header is included in PLAY_NOTIFY requests with the Notify-Reason of
the end of stream to provide RTP information about the end of the
stream.The header can carry the following parameters: Indicates the stream URI for which the
following RTP parameters correspond; this URI MUST be the same as
used in the SETUP request for this media stream. Any relative URI
MUST use the Request-URI as base URI. This parameter MUST be
present.The SSRC to which the RTP timestamp and
sequence number provided applies. This parameter MUST be
present.Indicates the sequence number of the first
packet of the stream that is direct result of the request. This
allows clients to gracefully deal with packets when seeking. The
client uses this value to differentiate packets that originated
before the seek from packets that originated after the seek. Note
that a client may not receive the packet with the expressed
sequence number and instead may receive packets with a higher
sequence number due to packet loss or reordering. This parameter
is RECOMMENDED to be present.MUST indicate the RTP timestamp value
corresponding to the start time value in the Range response-header
or, if not explicitly given, the implied start point. The client
uses this value to calculate the mapping of RTP time to NPT or
other media timescale. This parameter SHOULD be present to ensure
inter-media synchronization is achieved. There exists no
requirement that any received RTP packet will have the same RTP
timestamp value as the one in the parameter used to establish
synchronization.A mapping from RTP timestamps to NTP format timestamps
(wallclock) is available via RTCP. However, this information is
not sufficient to generate a mapping from RTP timestamps to media
clock time (NPT, etc.). Furthermore, in order to ensure that this
information is available at the necessary time (immediately at
startup or after a seek), and that it is delivered reliably, this
mapping is placed in the RTSP control channel.In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.Example:The Scale general-header indicates the requested or used view rate
for the media resource being played back. A scale value of 1 indicates
normal play at the normal forward viewing rate. If not 1, the value
corresponds to the rate with respect to normal viewing rate. For
example, a value of 2 indicates twice the normal viewing rate ("fast
forward") and a value of 0.5 indicates half the normal viewing rate.
In other words, a value of 2 has content time increase at twice the
playback time. For every second of elapsed (wallclock) time, 2 seconds
of content time will be delivered. A negative value indicates reverse
direction. For certain media transports, this may require certain
considerations to work consistently; see for
description on how RTP handles this.The transmitted-data rate SHOULD NOT be changed by selection of a
different scale value. The resulting bitrate should be reasonably
close to the nominal bitrate of the content for scale = 1. The server
has to actively manipulate the data when needed to meet the bitrate
constraints. Implementation of scale changes depends on the server and
media type. For video, a server may, for example, deliver only key
frames or selected frames. For audio, it may time-scale the audio
while preserving pitch or, less desirably, deliver fragments of audio,
or completely mute the audio.The server and content may restrict the range of scale values that
it supports. The supported values are indicated by the Media-Properties header. The
client SHOULD only indicate request values to be supported. However,
as the values may change as the content progresses, a requested value
may no longer be valid when the request arrives. Thus, a non-supported
value in a request does not generate an error, it only forces the
server to choose the closest value. The response MUST always contain
the actual scale value chosen by the server.If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting scale operations for
PLAY MUST indicate this with the use of the "play.scale" feature
tag.When indicating a negative scale for a reverse playback, the Range
header MUST indicate a decreasing range as described in .Example of playing in reverse at 3.5 times normal rate:When a client sends a PLAY request with a Range header to perform a
random access to the media, the client does not know if the server
will pick the first media samples or the first random access point
prior to the request range. Depending on the use case, the client may
have a strong preference. To express this preference and provide the
client with information on how the server actually acted on that
preference, the Seek-Style general-header is defined.Seek-Style is a general-header that MAY be included in any PLAY
request to indicate the client's preference for any media stream that
has the random access properties. The server MUST always include the
header in any PLAY response for media with random access properties to
indicate what policy was applied. A server that receives an unknown
Seek-Style policy MUST ignore it and select the server default policy.
A client receiving an unknown policy MUST ignore it and use the Range
header and any media synchronization information as basis to determine
what the server did.This specification defines the following seek policies that may be
requested (see also ):Random Access Point (RAP) is the behavior of
requesting the server to locate the closest previous random access
point that exists in the media aggregate and deliver from that. By
requesting a RAP, media quality will be the best possible as all
media will be delivered from a point where full media state can be
established in the media decoder.Conditional Random Access Point (CoRAP) is a
variant of the above RAP behavior. This policy is primarily
intended for cases where there is larger distance between the
random access points in the media. CoRAP uses the RAP policy if
the condition that there is a Random Access Point closer to the
requested start point than to the current pause point is
fulfilled. Otherwise, no seeking is performed and playback will
continue from the current pause point. This policy assumes that
the media state existing prior to the pause is usable if delivery
is continued. If the client or server knows that this is not the
fact, the RAP policy should be used. In other words, in most cases
when the client requests a start point prior to the current pause
point, a valid decoding dependency chain from the media delivered
prior to the pause and to the requested media unit will not exist.
If the server searched to a random access point, the server MUST
return the CoRAP policy in the Seek-Style header and adjust the
Range header to reflect the position of the selected RAP. In case
the random access point is farther away and the server chooses to
continue from the current pause point, it MUST include the "Next"
policy in the Seek-Style header and adjust the Range header start
point to the current pause point.The first-prior policy will start
delivery with the media unit that has a playout time first prior
to the requested time. For discrete media, that would only include
media units that would still be rendered at the request time. For
continuous media, that is media that will be rendered during the
requested start time of the range.The next media units after the provided start
time of the range: for continuous framed media, that would mean
the first next frame after the provided time and for discrete
media, the first unit that is to be rendered after the provided
time. The main usage for this case is when the client knows it has
all media up to a certain point and would like to continue
delivery so that a complete uninterrupted media playback can be
achieved. An example of such a scenario would be switching from a
broadcast/multicast delivery to a unicast-based delivery. This
policy MUST only be used on the client's explicit request.Please note that these expressed preferences exist for
optimizing the startup time or the media quality. The "Next" policy
breaks the normal definition of the Range header to enable a client to
request media with minimal overlap, although some may still occur for
aggregated sessions. RAP and First-Prior both fulfill the requirement
of providing media from the requested range and forward. However,
unless RAP is used, the media quality for many media codecs using
predictive methods can be severely degraded unless additional data is
available as, for example, already buffered, or through other side
channels.The Server general-header field contains information about the
software used by the origin server to create or handle the request.
This field can contain multiple product tokens and comments
identifying the server and any significant subproducts. The product
tokens are listed in order of their significance for identifying the
application.Example:If the response is being forwarded through a proxy, the proxy
application MUST NOT modify the Server response-header. Instead, it
SHOULD include a Via field. If the
response is generated by the proxy, the proxy application MUST return
the Server response-header as previously returned by the server.The Session general-header field identifies an RTSP session. An
RTSP session is created by the server as a result of a successful
SETUP request, and in the response, the session identifier is given to
the client. The RTSP session exists until destroyed by a TEARDOWN or a
REDIRECT or is timed out by the server.The session identifier is chosen by the server (see ) and MUST be returned in the SETUP response.
Once a client receives a session identifier, it MUST be included in
any request related to that session. This means that the Session
header MUST be included in a request, using the following methods:
PLAY, PAUSE, PLAY_NOTIFY and TEARDOWN. It MAY be included in SETUP,
OPTIONS, SET_PARAMETER, GET_PARAMETER, and REDIRECT. It MUST NOT be
included in DESCRIBE. The Session header MUST NOT be included in the
following methods, if these requests are pipelined and if the session
identifier is not yet known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS
SET_PARAMETER, and GET_PARAMETER.In an RTSP response, the session header MUST be included in
methods, SETUP, PLAY, PAUSE, and PLAY_NOTIFY, and it MAY be included
in methods TEARDOWN and REDIRECT. If included in the request of the
following methods it MUST also be included in the response: OPTIONS,
GET_PARAMETER, and SET_PARAMETER. It MUST NOT be included in DESCRIBE
responses.Note that a session identifier identifies an RTSP session across
transport sessions or connections. RTSP requests for a given session
can use different URIs (Presentation and media URIs). Note, that there
are restrictions depending on the session as to which URIs are
acceptable for a given method. However, multiple "user" sessions for
the same URI from the same client will require use of different
session identifiers.The session identifier is needed to distinguish several
delivery requests for the same URI coming from the same
client.The response 454 (Session Not Found) MUST be returned if the
session identifier is invalid.The header MAY include a parameter for session timeout period. If
not explicitly provided, this value is set to 60 seconds. As this
affects how often session keep-alives are needed, values smaller than
30 seconds are not recommended. However, larger-than-default values
can be useful in applications of RTSP that have inactive but
established sessions for longer time periods.The 60-second value was chosen as the session timeout value as
it results in keep-alive messages that are not too frequent and
low sensitivity to variations in request/response timing. If one
reduces the timeout value to below 30 seconds, the corresponding
request/response timeout becomes a significant part of the session
timeout. The 60-second value also allows for reasonably rapid
recovery of committed server resources in case of client
failure.The Speed general-header field requests the server to deliver
specific amounts of nominal media time per unit of delivery time,
contingent on the server's ability and desire to serve the media
stream at the given speed. The client requests the delivery speed to
be within a given range with a lower and upper bound. The server SHALL
deliver at the highest possible speed within the range, but not faster
than the upper bound, for which the underlying network path can
support the resulting transport data rates. As long as any speed value
within the given range can be provided, the server SHALL NOT modify
the media quality. Only if the server is unable to deliver media at
the speed value provided by the lower bound shall it reduce the media
quality.Implementation of the Speed functionality by the server is
OPTIONAL. The server can indicate its support through a feature tag,
play.speed. The lack of a Speed header in the response is an
indication of lack of support of this functionality.The speed parameter values are expressed as a positive decimal
value, e.g., a value of 2.0 indicates that data is to be delivered
twice as fast as normal. A speed value of zero is invalid. The range
is specified in the form "lower bound - upper bound". The lower-bound
value may be smaller or equal to the upper bound. All speeds may not
be possible to support. Therefore, the server MAY modify the requested
values to the closest supported. The actual supported speed MUST be
included in the response. However, note that the use cases may vary
and that Speed value ranges such as 0.7-0.8, 0.3-2.0, 1.0-2.5, and
2.5-2.5 all have their usages.Example:Use of this header changes the bandwidth used for data delivery. It
is meant for use in specific circumstances where delivery of the
presentation at a higher or lower rate is desired. The main use cases
are buffer operations or local scale operations. Implementers should
keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP) and, therefore, renegotiation
may be necessary. To perform Speed operations, the server needs to
ensure that the network path can support the resulting bitrate. Thus,
the media transport needs to support feedback so that the server can
react and adapt to the available bitrate.The Supported general-header enumerates all the extensions
supported by the client or server using feature tags. The header
carries the extensions supported by the message-sending client or
server. The Supported header MAY be included in any request. When
present in a request, the receiver MUST respond with its corresponding
Supported header. Note that the Supported header is also included in
4xx and 5xx responses.The Supported header contains a list of feature tags, described in
, that are understood by the client
or server. These feature tags are the ones the server or client
supports in general and are not specific to the request resource.Example:The Terminate-Reason request-header allows the server, when sending
a REDIRECT or TEARDOWN request, to provide a reason for the session
termination and any additional information. This specification
identifies three reasons for Redirections and may be extended in the
future:The server needs to be shut down for
some administrative reason.A client's session has been kept
alive for extended periods of time and the server has determined
that it needs to reclaim the resources associated with this
session.An internal error that is impossible
to recover from has occurred, forcing the server to terminate the
session.The Server may provide additional parameters containing
information around the redirect. This specification defines the
following ones.Provides a wallclock time when the server will
stop providing any service.A UTF-8 text string with a message from
the server to the user. This message SHOULD be displayed to the
user.The Timestamp general-header describes when the agent sent the
request. The value of the timestamp is of significance only to the
agent and may use any timescale. The responding agent MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating-point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp can be used
by the agent to compute the round-trip time to the responding agent so
that it can adjust the timeout value for retransmissions when running
over an unreliable protocol. It also resolves retransmission
ambiguities for unreliable transport of RTSP.Note that the present specification provides only for reliable
transport of RTSP messages. The Timestamp general-header is specified
in case the protocol is extended in the future to use unreliable
transport.The Transport general-header indicates which transport protocol is
to be used and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a presentation
description.A Transport request-header MAY contain a list of transport options
acceptable to the client, in the form of multiple transport
specification entries. Transport specifications are comma separated
and listed in decreasing order of preference. Each transport
specification consists of a transport protocol identifier, followed by
any number of parameters separated by semicolons. A Transport
request-header MAY contain multiple transport specifications using the
same transport protocol identifier. The server MUST return a Transport
response-header in the response to indicate the values actually
chosen, if any. If no transport specification is supported, no
transport header is returned and the response MUST use the status code
461 (Unsupported Transport). In
case more than one transport specification was present in the request,
the server MUST return the single transport specification
(transport-spec) that was actually chosen, if any. The number of
transport-spec entries is expected to be limited as the client will
receive guidance on what configurations are possible from the
presentation description.The Transport header MAY also be used in subsequent SETUP requests
to change transport parameters. A server MAY refuse to change
parameters of an existing stream.The transport protocol identifier defines, for each transport
specification, which transport protocol to use and any related rules.
Each transport protocol identifier defines the parameters that are
required to occur; additional optional parameters MAY occur. This
flexibility is provided as parameters may be different and provide
different options to the RTSP agent. A transport specification may
only contain one of any given parameter within it. A parameter
consists of a name and optionally a value string. Parameters MAY be
given in any order. Additionally, a transport specification may only
contain either the unicast or the multicast transport type parameter.
The transport protocol identifier, and all parameters, need to be
understood in a transport specification; if not, the transport
specification MUST be ignored. An RTSP proxy of any type that uses or
modifies the transport specification, e.g., access proxy or security
proxy, MUST remove specifications with unknown parameters before
forwarding the RTSP message. If that results in no remaining transport
specification, the proxy SHALL send a 461
(Unsupported Transport) response without any Transport
header.The Transport header is restricted to describing a single media
stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of
firewalls.The general syntax for the transport protocol identifier is a list
of slash-separated tokens:Which, for RTP transports, takes the form:The default value for the "lower-transport" parameters is specific
to the profile. For RTP/AVP, the default is UDP.There are two different methods for how to specify where the media
should be delivered for unicast transport: The presence of this parameter and its
values indicates the destination address or addresses (host
address and port pairs for IP flows) necessary for the media
transport.The lack of the dest_addr parameter
indicates that the server MUST send media to the same address from
which the RTSP messages originates.The choice of method for indicating where the media is to be
delivered depends on the use case. In some cases, the only allowed
method will be to use no explicit address indication and have the
server deliver media to the source of the RTSP messages.For multicast, there are several methods for specifying addresses,
but they are different in how they work compared with unicast:The address
and relevant parameters, like TTL (scope), for the actual
multicast group to deliver the media to. There are security implications with this
method that need to be addressed because an RTSP server can be
used as a DoS attacker on an existing multicast group.The
information included in the transport header can all be coming
from the session description, e.g., the SDP "c=" and "m=" lines.
This mitigates some of the security issues of the previous methods
as it is the session provider that picks the multicast group and
scope. The client MUST include the information if it is available
in the session description.The behavior when no explicit
multicast group is present in a request is not defined.An RTSP proxy will need to take care. If the media is not
desired to be routed through the proxy, the proxy will need to
introduce the destination indication.Below are the configuration parameters associated with transport:
General parameters: This parameter is a mutually
exclusive indication of whether unicast or multicast delivery will
be attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission need to indicate such capability by including two
full transport-specs with separate parameters for each.The number of multicast layers to be used
for this media stream. The layers are sent to consecutive
addresses starting at the dest_addr address. If the parameter is
not included, it defaults to a single layer.A general destination address parameter
that can contain one or more address specifications. Each
combination of protocol/profile/lower transport needs to have the
format and interpretation of its address specification defined.
For RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a
tuple containing a host address and port. Note, only a single
destination parameter per transport spec is intended. The usage of
multiple destinations to distribute a single media to multiple
entities is unspecified. The client
originating the RTSP request MAY specify the destination address
of the stream recipient with the host address as part of the
tuple. When the destination address is specified, the recipient
may be a different party than the originator of the request. To
avoid becoming the unwitting perpetrator of a remote-controlled
DoS attack, a server MUST perform security checks (see ) and SHOULD log such attempts before allowing
the client to direct a media stream to a recipient address not
chosen by the server. Implementations cannot rely on TCP as a
reliable means of client identification. If the server does not
allow the host address part of the tuple to be set, it MUST return
463 (Destination Prohibited). The host
address part of the tuple MAY be empty, for example ":58044", in
cases when it is desired to specify only the destination port.
Responses to requests including the Transport header with a
dest_addr parameter SHOULD include the full destination address
that is actually used by the server. The server MUST NOT remove
address information that is already present in the request when
responding, unless the protocol requires it.A general source address parameter that
can contain one or more address specifications. Each combination
of protocol/profile/lower transport needs to have the format and
interpretation of its address specification defined. For
RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
containing a host address and port. This
parameter MUST be specified by the server if it transmits media
packets from an address other than the one RTSP messages are sent
to. This will allow the client to verify the source address and
give it a destination address for its RTCP feedback packets, if
RTP is used. The address or addresses indicated in the src_addr
parameter SHOULD be used both for the sending and receiving of the
media stream's data packets. The main reasons are threefold:
First, indicating the port and source address(s) lets the receiver
know where from the packets is expected to originate. Second,
traversal of NATs is greatly simplified when traffic is flowing
symmetrically over a NAT binding. Third, certain NAT traversal
mechanisms need to know to which address and port to send
so-called "binding packets" from the receiver to the sender, thus
creating an address binding in the NAT that the sender-to-receiver
packet flow can use. This information may also be available through SDP.
However, since this is more a feature of transport than media
initialization, the authoritative source for this information
should be in the SETUP response.The mode parameter indicates the methods to be
supported for this session. The currently defined valid value is
"PLAY". If not provided, the default is "PLAY". The "RECORD" value
was defined in RFC 2326; in this specification, it is unspecified
but reserved. RECORD and other values may be specified in the
future.The interleaved parameter implies
mixing the media stream with the control stream in whatever
protocol is being used by the control stream, using the mechanism
defined in . The argument provides the
channel number to be used in the $ block (see ) and MUST be present. This parameter MAY be
specified as an interval, e.g., interleaved=4-5 in cases where the
transport choice for the media stream requires it, e.g., for RTP
with RTCP. The channel number given in the request is only a
guidance from the client to the server on what channel number(s)
to use. The server MAY set any valid channel number in the
response. The declared channels are bidirectional, so both end
parties MAY send data on the given channel. One example of such
usage is the second channel used for RTCP, where both server and
client send RTCP packets on the same channel. This allows RTP/RTCP to be handled similarly to the way
that it is done with UDP, i.e., one channel for RTP and the
other for RTCP.This parameter is used in conjunction with
transport specifications that can utilize MIKEY for security context establishment.
So far, only the SRTP-based RTP profiles SAVP and SAVPF can
utilize MIKEY, and this is defined in .
This parameter can be included both in request and response
messages. The binary MIKEY message SHALL be Base64-encoded before being included in
the value part of the parameter, where the encoding adheres to the
definition in Section 4 of RFC 4648 and where the padding bits are
set to zero.Multicast-specific: multicast time-to-live for IPv4. When included
in requests, the value indicates the TTL value that the client
requests the server to use. In a response, the value actually
being used by the server is returned. A server will need to
consider what values that are reasonable and also the authority of
the user to set this value. Corresponding functions are not needed
for IPv6 as the scoping is part of the IPv6
multicast address.RTP-specific: These parameters MAY only be
used if the media-transport protocol is RTP. The ssrc parameter, if included in a SETUP
response, indicates the RTP SSRC value(s)
that will be used by the media server for RTP packets within the
stream. The values are expressed as a slash-seperated sequence of
SSRC values, each SSRC expressed as an eight-digit hexadecimal
value. The ssrc parameter MUST NOT be
specified in requests. The functionality of specifying the ssrc
parameter in a SETUP request is deprecated as it is incompatible
with the specification of RTP. If
the parameter is included in the Transport header of a SETUP
request, the server SHOULD ignore it, and choose appropriate SSRCs
for the stream. The server SHOULD set the ssrc parameter in the
Transport header of the response.Used to negotiate the usage of RTP and RTCP multiplexing on a single
underlying transport stream/flow. The presence of this parameter
in a SETUP request indicates the client's support and requires the
server to use RTP and RTCP multiplexing. The client SHALL only
include one transport stream in the Transport header
specification. To provide the server with a choice between using
RTP/RTCP multiplexing or not, two different transport header
specifications must be included.The parameter setup and connection defined below MAY only be used
if the media-transport protocol of the lower-level transport is
connection oriented (such as TCP). However, these parameters MUST NOT
be used when interleaving data over the RTSP connection.Clients use the setup parameter on the
Transport line in a SETUP request to indicate the roles it wishes
to play in a TCP connection. This parameter is adapted from . The use of this parameter in RTP/AVP/TCP
non-interleaved transport is discussed in ; the discussion below is limited
to syntactic issues. Clients may specify the following values for
the setup parameter: The client will initiate an outgoing
connection.The client will accept an incoming
connection.The client is willing to accept an
incoming connection or to initiate an outgoing connection. If a client does not specify a
setup value, the "active" value is assumed. In response to a client SETUP request where the
setup parameter is set to "active", a server's 2xx reply MUST
assign the setup parameter to "passive" on the Transport header
line. In response to a client SETUP
request where the setup parameter is set to "passive", a server's
2xx reply MUST assign the setup parameter to "active" on the
Transport header line. In response to a
client SETUP request where the setup parameter is set to
"actpass", a server's 2xx reply MUST assign the setup parameter to
"active" or "passive" on the Transport header line. Note that the "holdconn" value for setup is not
defined for RTSP use, and MUST NOT appear on a Transport line.Clients use the connection parameter in
a transport specification part of the Transport header in a SETUP
request to indicate the client's preference for either reusing an
existing connection between client and server (in which case the
client sets the "connection" parameter to "existing") or
requesting the creation of a new connection between client and
server (in which cast the client sets the "connection" parameter
to "new"). Typically, clients use the "new" value for the first
SETUP request for a URL, and "existing" for subsequent SETUP
requests for a URL. If a client SETUP
request assigns the "new" value to "connection", the server
response MUST also assign the "new" value to "connection" on the
Transport line. If a client SETUP request
assigns the "existing" value to "connection", the server response
MUST assign a value of "existing" or "new" to "connection" on the
Transport line, at its discretion. The
default value of "connection" is "existing", for all SETUP
requests (initial and subsequent).The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the .Below is a usage example, showing a client advertising the
capability to handle multicast or unicast, preferring multicast. Since
this is a unicast-only stream, the server responds with the proper
transport parameters for unicast.The Unsupported response-header lists the features not supported by
the responding RTSP agent. In the case where the feature was specified
via the Proxy-Require field (), if
there is a proxy on the path between the client and the server, the
proxy MUST send a response message with a status code of 551 (Option
Not Supported). The request MUST NOT be forwarded.See for a usage example.The User-Agent general-header field contains information about the
user agent originating the request or producing a response. This is
for statistical purposes, the tracing of protocol violations, and
automated recognition of user agents for the sake of tailoring
responses to avoid particular user agent limitations. User agents
SHOULD include this field with requests. The field can contain
multiple product tokens and comments identifying the agent and any
subproducts which form a significant part of the user agent. By
convention, the product tokens are listed in order of their
significance for identifying the application.Example:The Via general-header field MUST be used by proxies to indicate
the intermediate protocols and recipients between the user agent and
the server on requests and between the origin server and the client on
responses. The field is intended to be used for tracking message
forwards, avoiding request loops, and identifying the protocol
capabilities of all senders along the request/response chain.Each of multiple values in the Via field represents each proxy that
has forwarded the message. Each recipient MUST append its information
such that the end result is ordered according to the sequence of
forwarding applications. So messages originating with the client or
server do not include the Via header. The first proxy or other
intermediate adds the header and its information into the field. Any
additional intermediate adds additional field-values. Resulting in the
server seeing the chains of intermediates in a client-to-server
request and the client seeing the full chain in the response
message.Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
default, forward the names and ports of hosts within the
private/protected region. This information SHOULD only be propagated
if explicitly enabled. If not enabled, the via-received of any host
behind the firewall/NAT SHOULD be replaced by an appropriate pseudonym
for that host.For organizations that have strong privacy requirements for hiding
internal structures, a proxy MAY combine an ordered subsequence of Via
header field entries with identical sent-protocol values into a single
such entry. Applications MUST NOT combine entries that have different
received-protocol values.The WWW-Authenticate header is specified in ; its usage depends on the used authentication
schemes, such as Digest and Basic. The WWW-Authenticate response-header
field MUST be included in 401 (Unauthorized) response messages. The
field-value consists of at least one challenge that indicates the
authentication scheme(s) and parameters applicable to the Request-URI.
This header MUST only be used in response messages related to client
to server requests.The HTTP access authentication process is described in with some clarification in . User agents are advised to
take special care in parsing the WWW-Authenticate field-value as it
might contain more than one challenge, or if more than one
WWW-Authenticate header field is provided, the contents of a challenge
itself can contain a comma-separated list of authentication
parameters.The RTSP security framework consists of two high-level components:
the pure authentication mechanisms based on HTTP authentication and the
message transport protection based on TLS, which is independent of RTSP.
Because of the similarity in syntax and usage between RTSP servers and
HTTP servers, the security for HTTP is reused to a large extent.RTSP and HTTP share common authentication schemes; thus, they
follow the same framework as specified in .
RTSP uses the corresponding RTSP error codes (401 and 407) and headers
(WWW-Authenticate, Authorization, Proxy-Authenticate,
Proxy-Authorization) by importing the definitions from . Servers SHOULD implement both the Basic and the Digest authentication schemes. Clients MUST
implement both the Basic and the Digest authentication schemes so that
a server that requires the client to authenticate can trust that the
capability is present. If implementing the Digest authentication
scheme, then the additional considerations specified below in MUST be followed.It should be stressed that using the HTTP authentication alone does
not provide full RTSP message security. Therefore, TLS SHOULD be used;
see . Any RTSP message containing an
Authorization header using the Basic authentication scheme MUST be
using a TLS connection with confidentiality protection enabled, i.e.,
no NULL encryption.In cases where there is a chain of proxies between the client and
the server, each proxy may individually request the client or previous
proxy to authenticate itself. This is done using the Proxy-Authenticate, the Proxy-Authorization, and the
Proxy-Authentication-Info
headers. These headers are hop-by-hop headers and are only scoped to
the current connection and hop. Thus, if a proxy chain exists, a proxy
connecting to another proxy will have to act as a client to authorize
itself towards the next proxy. The WWW-Authenticate, Authorization, and Authentication-Info headers
are end-to-end and MUST NOT be modified by proxies.This authentication mechanism works only for client-to-server
requests as currently defined. This leaves server-to-client request
outside of the context of TLS-based communication more vulnerable to
message-injection attacks on the client. Based on the server-to-client
methods that exist, the potential risks are various: hijacking
(REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY), or attacks
with uncertain results (SET_PARAMETER).This section describes the modifications and clarifications
required to apply the HTTP Digest authentication scheme to RTSP. The
RTSP scheme usage is almost completely identical to that for HTTP. These modifications are based on the
procedures defined for SIP 2.0 (in
Section 22.4) but updated to use RFC 7235, RFC 7616 and RFC 7615
instead of RFC 2617.Digest authentication uses two additional headers,
Authentication-Info and Proxy-Authentication-Info, that are defined
as in . The rules for Digest authentication
follow those defined in , with "HTTP/1.1"
replaced by "RTSP/2.0" in addition to the following
differences:Use the ABNF specified in the referenced documents, with the
difference that the URI parameter uses the request URI format
for RTSP, i.e. the ABNF element: Request-URI (see ). The domain parameter uses
the RTSP-URI-Ref element for absolute and relative URIs.If MTags are used, then the example procedure for choosing a
nonce based on ETag can work, based on replacing the ETag with
the MTag.As a clarification to the calculation of the A2 value for
message integrity assurance in the Digest authentication scheme,
implementers should assume, when the entity-body is empty (that
is, when the RTSP messages have no message body) that the hash
of the message body resolves to the hash of an empty string, or:
H(entity-body), example MD5("") =
"d41d8cd98f00b204e9800998ecf8427e".RTSP agents MUST implement RTSP over TLS as defined in this section
and the next . RTSP MUST follow
the same guidelines with regard to TLS usage
as specified for HTTP; see . RTSP over TLS is
separated from unsecured RTSP both on the URI level and the port
level. Instead of using the "rtsp" scheme identifier in the URI, the
"rtsps" scheme identifier MUST be used to signal RTSP over TLS. If no
port is given in a URI with the "rtsps" scheme, port 322 MUST be used
for TLS over TCP/IP.When a client tries to set up an insecure channel to the server
(using the "rtsp" URI), and the policy for the resource requires a
secure channel, the server MUST redirect the client to the secure
service by sending a 301 redirect response code together with the
correct Location URI (using the "rtsps" scheme). A user or client MAY
upgrade a non secured URI to a secured by changing the scheme from
"rtsp" to "rtsps". A server implementing support for "rtsps" MUST
allow this.It should be noted that TLS allows for mutual authentication (when
using both server and client certificates). Still, one of the more
common ways TLS is used is to provide only server-side authentication
(often to avoid client certificates). TLS is then used in addition to
HTTP authentication, providing transport security and server
authentication, while HTTP Authentication is used to authenticate the
client.RTSP includes the possibility to keep a TCP session up between the
client and server, throughout the RTSP session lifetime. It may be
convenient to keep the TCP session, not only to save the extra setup
time for TCP, but also the extra setup time for TLS (even if TLS uses
the resume function, there will be almost two extra round trips).
Still, when TLS is used, such behavior introduces extra active state
in the server, not only for TCP and RTSP, but also for TLS. This may
increase the vulnerability to DoS attacks.There exists a potential security vulnerability when reusing TCP
and TLS state for different resources (URIs). If two different
hostnames point at the same IP address, it can be desirable to reuse
the TCP/TLS connection to that server. In that case, the RTSP agent
having the TCP/TLS connection MUST verify that the server certificate
associated with the connection has a SubjectAltName matching the
hostname present in the URI for the resource an RTSP request is to be
issued.In addition to these recommendations, gives further recommendations of TLS
usage with proxies.The nature of a proxy is often to act as a "man in the middle",
while security is often about preventing the existence of one. This
section provides clients with the possibility to use proxies even when
applying secure transports (TLS) between the RTSP agents. The TLS
proxy mechanism allows for server and proxy identification using
certificates. However, the client cannot be identified based on
certificates. The client needs to select between using the procedure
specified below or using a TLS connection directly (bypassing any
proxies) to the server. The choice may be dependent on policies.In general, there are two categories of proxies: the transparent
proxies (of which the client is not aware) and the non-transparent
proxies (of which the client is aware). This memo specifies only
non-transparent RTSP proxies, i.e., proxies visible to the RTSP client
and RTSP server. An infrastructure based on proxies requires that the
trust model be such that both client and server can trust the proxies
to handle the RTSP messages correctly. To be able to trust a proxy,
the client and server also need to be aware of the proxy. Hence,
transparent proxies cannot generally be seen as trusted and will not
work well with security (unless they work only at the transport
layer). In the rest of this section, any reference to "proxy" will be
to a non-transparent proxy, which inspects or manipulates the RTSP
messages.HTTP Authentication is built on the assumption of proxies and can
provide user-proxy authentication and proxy-proxy/server
authentication in addition to the client-server authentication.When TLS is applied and a proxy is used, the client will connect to
the proxy's address when connecting to any RTSP server. This implies
that for TLS, the client will authenticate the proxy server and not
the end server. Note that when the client checks the server
certificate in TLS, it MUST check the proxy's identity (URI or
possibly other known identity) against the proxy's identity as
presented in the proxy's Certificate message.The problem is that for a proxy accepted by the client, the proxy
needs to be provided information on which grounds it should accept the
next-hop certificate. Both the proxy and the user may have rules for
this, and the user should have the possibility to select the desired
behavior. To handle this case, the Accept-Credentials header (see
) is used, where the client can
request the proxy or proxies to relay back the chain of certificates
used to authenticate any intermediate proxies as well as the server.
The assumption that the proxies are viewed as trusted gives the user a
possibility to enforce policies on each trusted proxy of whether it
should accept the next agent in the chain. However, it should be noted
that not all deployments will return the chain of certificates used to
authenticate any intermediate proxies as well as the server. An
operator of such a deployment may want to hide its topology from the
client. It should be noted well that the client does not have any
insight into the proxy's operation. Even if the proxy is trusted, it
can still return an incomplete chain of certificates.A proxy MUST use TLS for the next hop if the RTSP request includes
an "rtsps" URI. TLS MAY be applied on intermediate links (e.g.,
between client and proxy or between proxy and proxy) even if the
resource and the end server are not required to use it. The chain of
proxies used by a client to reach a server and its TLS sessions MUST
have commensurate security. Therefore, a proxy MUST, when initiating
the next-hop TLS connection, use the incoming TLS connections
cipher-suite list, only modified by removing any cipher suites that
the proxy does not support. In case a proxy fails to establish a TLS
connection due to cipher-suite mismatch between proxy and next-hop
proxy or server, this is indicated using error code 472 (Failure to
Establish Secure Connection).The Accept-Credentials header can be used by the client to
distribute simple authorization policies to intermediate proxies.
The client includes the Accept-Credentials header to dictate how the
proxy treats the server / next proxy certificate. There are
currently three methods defined: With "any", the proxy (or proxies) MUST
accept whatever certificate is presented. Of course, this is not
a recommended option to use, but it may be useful in certain
circumstances (such as testing).For the "proxy" method, the proxy (or
proxies) MUST use its own policies to validate the certificate
and decide whether or not to accept it. This is convenient in
cases where the user has a strong trust relation with the proxy.
Reasons why a strong trust relation may exist are
personal/company proxy or the proxy has an out-of-band policy
configuration mechanism.For the "user" method, the proxy (or
proxies) MUST send credential information about the next hop to
the client for authorization. The client can then decide whether
or not the proxy should accept the certificate. See for further details.If the Accept-Credentials header is not included in the RTSP
request from the client, then the "Proxy" method MUST be used as
default. If a method other than the "Proxy" is to be used, then the
Accept-Credentials header MUST be included in all of the RTSP
requests from the client. This is because it cannot be assumed that
the proxy always keeps the TLS state or the user's previous
preference between different RTSP messages (in particular, if the
time interval between the messages is long).With the "Any" and "Proxy" methods, the proxy will apply the
policy as defined for each method. If the policy does not accept the
credentials of the next hop, the proxy MUST respond with a message
using status code 471 (Connection Credentials Not Accepted).An RTSP request in the direction server to client MUST NOT
include the Accept-Credentials header. As for the non-secured
communication, the possibility for these requests depends on the
presence of a client established connection. However, if the
server-to-client request is in relation to a session established
over a TLS secured channel, it MUST be sent in a TLS secured
connection. That secured connection MUST also be the one used by the
last client-to-server request. If no such transport connection
exists at the time when the server desires to send the request, the
server MUST discard the message.Further policies MAY be defined and registered, but this should
be done with caution.For the "User" method, each proxy MUST perform the following
procedure for each RTSP request: Set up the TLS session to the next hop if not already present
(i.e., run the TLS handshake, but do not send the RTSP
request).Extract the peer certificate chain for the TLS session.Check if a matching identity and hash of the peer certificate
are present in the Accept-Credentials header. If present, send
the message to the next hop and conclude these procedures. If
not, go to the next step.The proxy responds to the RTSP request with a 470 or 407
response code. The 407 response code MAY be used when the proxy
requires both user and connection authorization from user or
client. In this message the proxy MUST include a
Connection-Credentials header, see , with the next hop's
identity and certificate.The client MUST upon receiving a 470 (Connection Authorization
Required) or 407 (Proxy Authentication Required) response with
Connection-Credentials header take the decision on whether or not to
accept the certificate (if it cannot do so, the user SHOULD be
consulted). Using IP addresses in the next-hop URI and certificates
rather than domain names makes it very difficult for a user to
determine whether or not it should approve the next hop. Proxies are
RECOMMENDED to use domain names to identify themselves in URIs and
in the certificates. If the certificate is accepted, the client has
to again send the RTSP request. In that request, the client has to
include the Accept-Credentials header including the hash over the
DER-encoded certificate for all trusted proxies in the chain.Example:One implication of this process is that the connection for
secured RTSP messages may take significantly more round-trip times
for the first message. A complete extra message exchange between the
proxy connecting to the next hop and the client results because of
the process for approval for each hop. However, if each message
contains the chain of proxies that the requester accepts, the
remaining message exchange should not be delayed. The procedure of
including the credentials in each request rather than building state
in each proxy avoids the need for revocation procedures.The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
as defined in RFC 5234 . It uses the basic
definitions present in RFC 5234.Please note that ABNF strings, e.g., "Accept", are case insensitive
as specified in Section 2.3 of RFC 5234.The RTSP syntax makes use of the ISO 10646 character set in UTF-8
encoding .RTSP header values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
whitespace, including folding, has the same semantics as SP. A
recipient MAY replace any linear whitespace with a single SP before
interpreting the field-value or forwarding the message downstream. The
SWS construct is used when linear whitespace is optional, generally
between tokens and separators.To separate the header name from the rest of value, a colon is
used, which, by the above rule, allows whitespace before, but no line
break, and whitespace after, including a line break. The HCOLON
defines this construct. This section defines in ABNF the SDP extensions defined for RTSP.
See for the definition of the extensions
in text.The security considerations and threats around RTSP and its usage can
be divided into considerations around the signaling protocol itself and
the issues related to the media-stream delivery. However, when it comes
to mitigation of security threats, a threat depending on the
media-stream delivery may in fact be mitigated by a mechanism in the
signaling protocol.There are several chapters and an appendix in this document that
define security solutions for the protocol. These sections will be
referenced when discussing the threats below. However, the reader should
take special notice of the Security Framework and the
specification of how to use SRTP and its
key-management to achieve certain aspects of the media
security.This section focuses on issues related to the signaling protocol.
Because of the similarity in syntax and usage between RTSP servers and
HTTP servers, the security considerations outlined in , , , , , and apply as well.Specifically, please note the following: A server is in the
position to save personal data about a user's requests that might
identify their media consumption patterns or subjects of interest.
This information is clearly confidential in nature, and its
handling can be constrained by law in certain countries. Log
information needs to be securely stored and appropriate guidelines
followed for its analysis. See Section 9.8 of for additional guidelines.There is no
reason to believe that information transferred in RTSP message,
such as the URI and the content of headers, especially the Server,
Via, Referrer, and From headers, may be any less sensitive than
when used in HTTP. Therefore, all of the precautions regarding the
protection of data privacy and user privacy apply to implementers
of RTSP clients, servers, and proxies. See Sections 9.3-9.6 of
for further details.The RTSP methods defined in this document are
primarily used to establish and control the delivery of the media
data represented by the URI; thus, the RTSP message bodies are
generally less sensitive than the ones in HTTP. Where HTTP bodies
could contain, for example, your medical records, in RTSP, the
sensitive video of your medical operation would be in the media
stream over the media-transport protocol, not in the RTSP message.
Still, one has to take note of what potential sensitive
information is included in RTSP. The protection of the media data
is separate, can be applied directly between client and server,
and is dependent on the media-transport protocol in use. See for further discussion. This
possibility for separation of security between media-resource
content and the signaling protocol mitigates the risk of exposing
the media content when using hop-by-hop security for RTSP signaling using
proxies.Though RTSP
URIs are opaque handles that do not necessarily have file-system
semantics, it is anticipated that many implementations will
translate portions of the Request-URIs directly to file-system
calls. In such cases, file systems SHOULD follow the precautions
outlined in Section 9.1 of , such as
checking for ".." in path components.RTSP clients are often privy
to the same information that HTTP clients are (username, location,
etc.) and thus should be equally sensitive. See Section 9.8 of
, Sections 9.3-9.7 of , and Section 8 of for
further recommendations.Since
similar usages of the "Accept" headers exist in RTSP as in HTTP,
the same caveats outlined in Section 9.4 of with regard to their use should be
followed.RTSP shares with HTTP the
question of how a client communicates with the authoritative
source for media streams (Section 9.1 of ). The used DNS servers, the security of the
communication, and any possibility of a man in the middle, and the
trust in any RTSP proxies all affect the possibility that a client
has received a non-authoritative response to a request. Ensuring
that a client receives an authoritative response is challenging,
although using the secure communication for RTSP signaling (rtsps)
simplifies it significantly as the server can provide a hostname
identity assertion in the TLS handshake.If a single server
supports multiple organizations that do not trust each another,
then it MUST check the values of the Content-Location header
fields in responses that are generated under control of said
organizations to make sure that they do not attempt to invalidate
resources over which they have no authority (see Section 15.4 of
).In addition to the recommendations in the current HTTP
specifications (, ,
, , , and as of this writing)
and also those of the previous relevant RFCs , future HTTP specifications may provide
additional guidance on security issues.The following are added considerations for RTSP implementations.
Since there is no or little
relation between a transport-layer connection and an RTSP session,
it is possible for a malicious client to issue requests with
random session identifiers that could affect other clients of an
unsuspecting server. To mitigate this, the server SHALL use a
large, random and non-sequential session identifier to minimize
the possibility of this kind of attack. However, unless the RTSP
signaling is always confidentiality protected, e.g., using TLS, an
on-path attacker will be able to hijack a session. Another choice
for preventing session hijacking is to use client authentication
and only allow the authenticated client creating the session to
access that session.Servers SHOULD implement both basic
and Digest authentication. In
environments requiring tighter security for the control messages,
the transport-layer mechanism TLS
SHOULD be used.Upon detecting instances of
behavior that is deemed a security risk, RTSP servers SHOULD
return error code 403 (Forbidden). RTSP servers SHOULD also be
aware of attempts to probe the server for weaknesses and entry
points and MAY arbitrarily disconnect and ignore further requests
from clients that are deemed to be in violation of local security
policy.If one uses the possibility to
connect TLS in multiple legs (), one really needs to be aware of
the trust model. This procedure requires trust in all proxies part
of the path to the server. The proxies one connects through are
identified, assuming the proxies so far connected through are well
behaved and fulfilling the trust. The accepted proxies are men in
the middle and have access to all that goes on over the TLS
connection. Thus, it is important to consider if that trust model
is acceptable in the actual application. Further discussion of the
actual trust model is in . It
is important to note what difference in security properties, if
any, may exist with the used media-transport protocol and its
security mechanism. Using SRTP and the MIKEY-based
key-establishment defined in enables
media key-establishment to be done end-to-end without revealing
the keys to the proxies.As RTSP is a stateful protocol
and establishes resource usage on the server, there is a clear
possibility to attack the server by trying to overbook these
resources to perform a DoS attack. This attack can be both against
ongoing sessions and to prevent others from establishing sessions.
RTSP agents will need to have mechanisms to prevent single peers
from consuming extensive amounts of resources. The methods for
guarding against this are varied and depend on the agent's role
and capabilities and policies. Each implementation has to
carefully consider its methods and policies to mitigate this
threat. There are recommendations regarding the handling of
connections in .The above threats and considerations have resulted in a set
of security functions and mechanisms built into or used by the
protocol. The signaling protocol relies on two security features
defined in the Security
Framework: namely client authentication using HTTP
authentication and TLS-based transport protection of the signaling
messages. Both of these mechanisms are required to be implemented by
any RTSP agent.A number of different security mitigations have been designed into
the protocol and will be instantiated if the specification is
implemented as written, for example, by ensuring sufficient amounts of
entropy in the randomly generated session identifiers when not using
client authentication to minimize the risk of session hijacking. When
client authentication is used, protection against hijacking will be
greatly improved by scoping the accessible sessions to the one this
client identity has created. Some of the above threats are such that
the implementation of the RTSP functionality itself needs to consider
which policy and strategy it uses to mitigate them.The fact that RTSP establishes and controls a media-stream delivery
results in a set of security issues related to the media streams. This
section will attempt to analyze general threats; however, the choice
of media-stream transport protocol, such as RTP, will result in some
differences in threats and what mechanisms exist to mitigate them.
Thus, it becomes important that each specification of a new
media-stream transport and delivery protocol usable by RTSP requires
its own security analysis. This section includes one for RTP.The set of general threats from or by the media-stream delivery
itself are:The protocol
offers the opportunity for a remote-controlled DoS attack, where
the media stream is the hammer in that DoS attack. See .The media delivery may
contain content of any type, and it is not possible, in general,
to determine how sensitive this content is from a confidentiality
point. Thus, it is a strong requirement that any media delivery
protocol supply a method for providing confidentiality of the
actual media content. In addition to the media-level
confidentiality, it becomes critical that no resource identifiers
used in the signaling be exposed to an attacker as they may have
human-understandable names or may be available to the attacker,
allowing it to determine the content the user received. Thus, the
signaling protocol must also provide confidentiality protection of
any information related to the media resource.There are
several reasons why an attacker will be interested in substituting
the media stream sent out from the RTSP server with one of the
attacker's creation or selection, such as discrediting the target
and misinformation about the target. Therefore, it is important
that the media protocol provide mechanisms to verify the source
authentication and integrity and to prevent replay attacks on the
media stream.If RTSP is used to control the
transmission of media onto a multicast network, the scope of the
delivery must be considered. RTSP supports the TTL Transport
header parameter to indicate this scope for IPv4. IPv6 has a
different mechanism for the scope boundary. However, such scope
control has risks, as it may be set too large and distribute media
beyond the intended scope.Below a protocol-specific
analysis of security considerations for RTP-based media transport is
included. In that section, the requirements on implementing security
functions for RTSP agents supporting media delivery over RTP are made
clear.The attacker may initiate traffic flows to one or more IP
addresses by specifying them as the destination in SETUP requests.
While the attacker's IP address may be known in this case, this is
not always useful in the prevention of more attacks or ascertaining
the attacker's identity. Thus, an RTSP server MUST only allow
client-specified destinations for RTSP-initiated traffic flows if
the server has ensured that the specified destination address
accepts receiving media through different security mechanisms.
Security mechanisms that are acceptable in order of increasing
generality are: Verification of the client's identity against a database of
known users using RTSP authentication mechanisms (preferably
Digest authentication or stronger)A list of addresses that have consented to be media
destinations, especially considering user identityVerification based on media pathThe server SHOULD NOT allow the destination field to be set
unless a mechanism exists in the system to authorize the request
originator to direct streams to the recipient. It is preferred that
this authorization be performed by the media recipient (destination)
itself and the credentials be passed along to the server. However,
in certain cases, such as when the recipient address is a multicast
group or when the recipient is unable to communicate with the server
in an out-of-band manner, this may not be possible. In these cases,
the server may choose another method such as a server-resident
authorization list to ensure that the request originator has the
proper credentials to request stream delivery to the recipient.One solution that performs the necessary verification of
acceptance of media suitable for unicast-based delivery is the NAT
traversal method based on Interactive
Connectivity Establishment (ICE) described in . This mechanism uses random passwords and a
username so that the probability of unintended indication as a valid
media destination is very low. In addition, the server includes in
its Session Traversal Utilities for NAT
(STUN) requests a cookie (consisting of random material) that
the destination echoes back; thus, the solution also safeguards
against having an off-path attacker being able to spoof the STUN
checks. This leaves this solution vulnerable only to on-path
attackers that can see the STUN requests go to the target of attack
and thus forge a response.For delivery to multicast addresses, there is a need for another
solution that is not specified in this memo.RTP is a commonly used media-transport protocol and has been the
most common choice for RTSP 1.0 implementations. The core RTP
protocol has been in use for a long time, and it has well-known
security properties and the RTP security consideration (Section 9 of
) needs to be reviewed. In perspective of
the usage of RTP in the context of RTSP, the following properties
should be noted:RTP has support for multiple
simultaneous media streams in each RTP session. As some use
cases require support for non-synchronized adding and removal of
media streams and their identifiers, an attacker can easily
insert additional media streams into a session context that,
according to protocol design, is intended to be played out.
Another threat vector is one of DoS by exhausting the resources
of the RTP session receiver, for example, by using a large
number of SSRC identifiers simultaneously. The strong mitigation
of this is to ensure that one cryptographically authenticates
any incoming packet flow to the RTP session. Weak mitigations
like blocking additional media streams in session contexts
easily lead to a DoS vulnerability in addition to preventing
certain RTP extensions or use cases that rely on multiple media
streams, such as RTP
retransmission to function.The built-in RTCP also offers a
large attack surface for a couple of different types of attacks.
One venue is to send RTCP feedback to the media sender
indicating large amounts of packet loss and thus trigger a media
bitrate adaptation response from the sender resulting in lowered
media quality and potentially a shutdown of the media stream.
Another attack is to perform a resource-exhaustion attack on the
receiver by using many SSRC identifiers to create large state
tables and increase the RTCP-related processing demands.RTP and RTCP extensions
generally provide additional and sometimes extremely powerful
tools for DoS attacks or service disruption. For example, the
Code Control Message RTCP
extensions enables both the lock down of the bitrate to low
values and disruption of video quality by requesting
intra-frames.Taking into account the above general discussion in and the RTP-specific discussion in
this section, it is clear that it is necessary that a strong
security mechanism be supported to protect RTP. Therefore, this
specification has the following requirements on RTP security
functions for all RTSP agents that handle media streams and where
media-stream transport is completed using RTP.RTSP agents supporting RTP MUST implement
Secure RTP (SRTP) and, thus, SAVP. In addition, SAVPF MUST also be supported if AVPF is implemented.
This specification requires no additional cryptographic transforms
or configuration values beyond those specified as mandatory to
implement in RFC 3711, i.e., AES-CM and HMAC-SHA1. The default
key-management mechanism that MUST be implemented is the one defined
in MIKEY Key Establishment. The
MIKEY implementation MUST implement the necessary functions for
MIKEY-RSA-R mode and the SRTP
parameter negotiation necessary to negotiate the supported SRTP
transforms and parameters.This section describes a number of registries for RTSP 2.0 that have
been established and are maintained by IANA. These registries are
separate from any registries existing for RTSP 1.0. For each
registry, there is a description of the required content, the
registration procedures, and the entries that this document
registers. For more information on extending RTSP, see . In addition,
this document registers three SDP attributes.Registries or entries in registries that have been made for RTSP 1.0
are not moved to RTSP 2.0: the registries and entries of RTSP 1.0 and
RTSP 2.0 are independent. If any registry or entry in a registry is also
required in RTSP 2.0, it MUST follow the procedure defined below to
allocate the registry or entry in a registry.The sections describing how to register an item use some of the
registration policies described in -- namely,
"First Come First Served", "Expert Review", "Specification Required",
and "Standards Action".In case a registry requires a contact person, the authors (with
Magnus Westerlund <magnus.westerlund@ericsson.com> as primary) are
the contact persons for any entries created by this document.IANA will request the following information for any registration
request: A name of the item to register according to the rules specified
by the intended registryIndication of who has change control over the feature (for
example, the IETF, ISO, ITU-T, other international standardization
bodies, a consortium, a particular company or group of companies, or
an individual)A reference to a further description, if available, for example
(in decreasing order of preference), an RFC, a published standard, a
published paper, a patent filing, a technical report, documented
source code or a computer manualFor proprietary features, contact information (postal and email
address)When a client and server try to determine what part and
functionality of the RTSP specification and any future extensions
that its counterpart implements, there is need for a namespace. This
registry contains named entries representing certain
functionality.The usage of feature tags is explained in and .The registering of feature tags is done on a First Come, First Served basis.The registry entry for a feature tag has the following
information:The name of the feature tagIf the registrant indicates that the feature is
proprietary, IANA should request a vendor "prefix" portion
of the name. The name will then be the vendor prefix
followed by a "." followed by the rest of the provided
feature name.If the feature is not proprietary, then IANA need not
collect a prefix for the name.A one-paragraph description of what the feature tag
representsThe applicability (server, client, proxy, or some
combination)A reference to a specification, if applicableFeature tag names (including the vendor prefix) may contain any
non-space and non-control characters. There is no length limit on
feature tags.Examples for a vendor tag describing a proprietary feature are:
vendorA.specfeat01vendorA.specfeat02The following feature tags are defined in this specification and
hereby registered. The change control belongs to the IETF. The implementation for delivery and
playback operations according to the core RTSP specification, as
defined in this memo. Applies for clients, servers, and proxies.
See .Support of scale operations for media
playback. Applies only for servers. See .Support of the speed functionality for
media delivery. Applies only for servers. See .Support of the RTP and RTCP
multiplexing as discussed in .
Applies for both client and servers and any media caching
proxy.The IANA registry is a table with the name, description,
and reference for each feature tag.Methods are described in . Extending
the protocol with new methods allows for totally new
functionality.A new method is registered through a
Standards Action because new methods may radically change the
protocol's behavior and purpose.A specification for a new RTSP method consists of the following
items: A method name that follows the ABNF rules for methods.A clear specification of what a request using the method does
and what responses are expected. In which directions the method
is used: C->S, S->C, or both. How the use of headers, if
any, modifies the behavior and effect of the method.A list or table specifying which of the IANA-registered
headers that are allowed to be used with the method in the
request or/and response. The list or table SHOULD follow the
format of tables in .Describe how the method relates to network proxies.This specification, RFC 7826, registers 10 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,
SET_PARAMETER, and TEARDOWN. The initial table of the registry is
provided below.A status code is the three-digit number used to convey
information in RTSP response messages; see . The number space is limited, and care
should be taken not to fill the space.A new status code registration follows the policy of IETF Review. New RTSP functionality
requiring Status Codes should first be registered in the range of
x50-x99. Only when the range is full should registrations be made in
the x00-x49 range, unless it is to adopt an HTTP extension to RTSP.
This is done to enable any HTTP extension to be adopted to RTSP
without needing to renumber any related status codes. A
specification for a new status code must include the following:
The registered number.A description of what the status code means and the expected
behavior of the sender and receiver of the code.RFC 7826 (this document) registers the numbered status code
defined in the ABNF entry "Status-Code", except "extension-code"
(that defines the syntax allowed for future extensions) in .By specifying new headers, one or more methods can be enhanced in
many different ways. An unknown header will be ignored by the
receiving agent. If the new header is vital for certain
functionality, a feature tag for the functionality can be created
and demanded to be used by the counterpart with the inclusion of a
Require header carrying the feature tag.Registrations can be made following the Expert Review policy. A specification is
recommended to be provided, preferably an RFC or other specification
from a Standards Developing Organization. The minimal information in
a registration request is the header name and the contact
information.The expert reviewer verifies that the registration request
contains the following information: The name of the header.An ABNF specification of the header syntax.A list or table specifying when the header may be used,
encompassing all methods, their request or response, and the
direction (C->S or S->C).How the header is to be handled by proxies.A description of the purpose of the header.All headers specified in in RFC 7826
have been registered. The registry includes the header name and
reference.Furthermore, the following legacy RTSP headers defined in other
specifications are registered with header name, and reference
according to below list. Note: these references may not fulfill all
of the above rules for registrations due to their legacy status.
x-wap-profile defined in . The
x-wap-profile request-header contains one or more absolute URLs
to the requesting agent's device-capability profile.x-wap-profile-diff defined in . The
x-wap-profile-diff request-header contains a subset of a
device-capability profile.x-wap-profile-warning defined in .
The x-wap-profile-warning is a response-header that contains
error codes explaining to what extent the server has been able
to match the terminal request in regard to device-capability
profiles, as described using x-wap-profile and
x-wap-profile-diff headers.x-predecbufsize defined in . This
response-header provides an RTSP agent with the TS 26.234 Annex
G hypothetical pre-decoder buffer size.x-initpredecbufperiod defined in .
This response-header provides an RTSP agent with the TS 26.234
Annex G hypothetical pre-decoder buffering period.x-initpostdecbufperiod defined in .
This response-header provides an RTSP agent with the TS 26.234
Annex G post-decoder buffering period.3gpp-videopostdecbufsize defined in . This response-header provides an RTSP agent
with the TS 26.234 defined post-decoder buffer size usable for
H.264 (AVC) video streams.3GPP-Link-Char defined in . This
request-header provides the RTSP server with the RTSP client's
link characteristics as determined from the radio interface. The
information that can be provided are guaranteed bitrate, maximum
bitrate and maximum transfer delay.3GPP-Adaptation defined in . This
general-header is part of the bitrate adaptation solution
specified for the Packet-switched Streaming Service (PSS). It
provides the RTSP client's buffer sizes and target buffer levels
to the server, and responses are used to acknowledge the support
and values.3GPP-QoE-Metrics defined in . This
general-header is used by PSS RTSP agents to negotiate the
quality of experience metrics that a client should gather and
report to the server.3GPP-QoE-Feedback defined in . This
request-header is used by RTSP clients supporting PSS to report
the actual values of the metrics gathered in its quality of
experience metering.The use of "x-" is NOT RECOMMENDED, but the above headers in the
list were defined prior to the clarification.The security framework's TLS connection mechanism has two
registerable entities.This registry is for policies for an RTSP proxy's handling and
verification of TLS certificates when establishing an outbound TLS
connection on behalf of a client. In , three policies for how to handle
certificates are specified. Further policies may be defined;
registration is made through Standards
Action. A registration request is required to contain the
following information: Name of the policy.Text that describes how the policy works for handling the
certificates.A contact person.This specification registers the following values: A policy requiring the proxy to accept any
received certificate.A policy where the proxy applies its own
policies to determine which certificates are accepted.A policy where the certificate is required
to be forwarded down the proxy chain to the client, thus
allowing the user to decided to accept or refuse a
certificate.The Accept-Credentials header (see ) allows for the usage of other
algorithms for hashing the DER records of accepted entities. The
registration of any future algorithm is expected to be extremely
rare and could also cause interoperability problems. Therefore, the
bar for registering new algorithms is intentionally placed high.Any registration of a new hash algorithm requires Standards Action. The registration needs to
fulfill the following requirement: The algorithms identifier meeting the "token" ABNF
requirement.Provide a definition of the algorithm.The registered value is:There exist a number of cache directives that can be sent in the
Cache-Control header. A registry for these cache directives has been
established by IANA. New registrations in this registry require Standards Action or IESG Approval. A
registration request needs to contain the following information. The name of the cache directive.A definition of the parameter value, if any is allowed.The specification if it is a request or response directive.Text that explains how the cache directive is used for
RTSP-controlled media streams.A contact person.This specification registers the following values: no-cache:public:private:no-transform:only-if-cached:max-stale:min-fresh:must-revalidate:proxy-revalidate:max-age:The registry contains the name of the directive and the
reference.The media streams being controlled by RTSP can have many
different properties. The media properties required to cover the use
cases that were in mind when writing the specification are defined.
However, it can be expected that further innovation will result in
new use cases or media streams with properties not covered by the
ones specified here. Thus, new media properties can be specified. As
new media properties may need a substantial amount of new
definitions to correctly specify behavior for this property, the bar
is intended to be high.Registering a new media property is done following the Specification Required policy. The expert
reviewer verifies that a registration request fulfills the following
requirements.An ABNF definition of the media property value name that
meets "media-prop-ext" definition is included.A definition of which media property group it belongs to or
define a new group is included.A description of all changes to the behavior of RTSP as
result of these changes is included.A contact person for the registration is listed.This specification registers the ten values listed in . The registry contains the property
group, the name of the media property, and the reference.Notify-Reason values are used to indicate the reason the
notification was sent. Each reason has its associated rules on what
headers and information may or must be included in the notification.
New notification behaviors need to be specified to enable
interoperable usage; thus, a specification of each new value is
required.Registrations for new Notify-Reason values follow the Specification Required policy. The expert
reviewer verifies that the request fulfills the following
requirements:An ABNF definition of the Notify-Reason value name that meets
"Notify-Reason-extension" definition is included.A description of which headers shall be included in the
request and response, when it should be sent, and any effect it
has on the server client state is made clear.A contact person for the registration is listed.This specification registers three values defined in the
Notify-Reas-val ABNF, :This Notify-Reason value indicates
the end of a media stream.This Notify-Reason value
allows the server to indicate that the properties of the media
have changed during the playout.This Notify-Reason value allows the
server to notify the client about a change in the scale of the
media.The registry contains the name, description, and
reference.The Range header allows for
different range formats. These range formats also need an identifier
to be used in the Accept-Ranges
header. New range formats may be registered, but moderation
should be applied as it makes interoperability more difficult.A registration follows the Specification
Required policy. The expert reviewer verifies that the
request fulfills the following requirements: An ABNF definition of the range format that fulfills the
"range-ext" definition is included.The range format identifier used in Accept-Ranges header
according to the "extension-format" definition is defined.Rules for how one handles the range when using a negative
Scale are included.A contact person for the registration is listed.The registry contains the Range header format identifier, the
name of the range format, and the reference. This specification
registers the following values.Normal Play TimeUTC Absolute Time formatSMPTE TimestampsSMPTE Timestamps 29.97 Frames/sec
(30 Hz with Drop)SMPTE Timestamps 25 Frames/secThe Terminate-Reason
header has two registries for extensions.This registry contains reasons for session termination that can
be included in a Terminate-Reason header.
Registrations follow the Expert Review
policy. The expert reviewer verifies that the registration
request contains the following information:That the value follows the Terminate-Reason ABNF, i.e., be a
token.That the specification provide a definition of what
procedures are to be followed when a client receives this
redirect reason.A contact personThis specification registers three values:Session-TimeoutServer-AdminInternal-ErrorThe registry contains the name of the Redirect Reason and
the reference.This registry contains parameters that may be included in the
Terminate-Reason header
in addition to a reason. Registrations are made under the Specification Required policy. The expert
reviewer verifies that the registration request contains the
following:A parameter name.A parameter following the syntax allowed by the RTSP 2.0
specification.A reference to the specification.A contact person. This specification registers:timeuser-msgThe registry contains the name of the Terminate Reason and
the reference.The RTP-Info header carries
one or more parameter value pairs with information about a
particular point in the RTP stream. RTP extensions or new usages may
need new types of information. As RTP information that could be
needed is likely to be generic enough, and to maximize the
interoperability, new registration is made under the Specification
Required policy.Registrations for new RTP-Info values follow the policy of Specification Required. The expert reviewer
verifies that the registration request contains the following
information.An ABNF definition that meets the "generic-param"
definition.A reference to the specification.A contact person for the registration.This specification registers the following parameter value
pairs:urlssrcseqrtptimeThe registry contains the name of the parameter and the
reference.Seek-Style policy defines how the RTSP agent seeks in media
content when given a position within the media content. New seek
policies may be registered; however, a large number of these will
complicate implementation substantially. The impact of unknown
policies is that the server will not honor the unknown and will use
the server default policy instead.Registrations of new Seek-Style policies follow the Specification Required policy. The expert
reviewer verifies that the registration request fulfills the
following requirements:Has an ABNF definition of the Seek-Style policy name that
meets "Seek-S-value-ext" definition.Includes a short description.Lists a contact person for the registration.Includes a description of which headers shall be included in
the request and response, when it should be sent, and any affect
it has on the server-client state.This specification registers four values (Name - Short
Description):RAP - Using the closest Random Access Point prior to or at
the requested start position.CoRAP - Conditional Random Access Point is like RAP, but only
if the RAP is closer prior to the requested start position than
current pause point.First-Prior - The first-prior policy will start delivery with
the media unit that has a playout time first prior to the
requested start position.Next - The next media units after the provided start
position.The registry contains the name of the Seek-Style policy,
the description, and the reference.The transport header contains a
number of parameters that have possibilities for future extensions.
Therefore, registries for these are defined below.A Transport Protocol specification consists of a transport
protocol identifier, representing some combination of transport
protocols, and any number of transport header parameters required or
optional to use with the identified protocol specification. This
registry contains the identifiers used by registered transport
protocol identifiers.A registration for the parameter transport protocol identifier
follows the Specification Required
policy. The expert reviewer verifies that the registration
request fulfills the following requirements: A contact person or organization with address and email.A value definition that follows the ABNF syntax definition of
"transport-id" .A descriptive text that explains how the registered values
are used in RTSP, which underlying transport protocols are used,
and any required Transport header parameters.The registry contains the protocol ID string and the
reference.This specification registers the following values: Use of the RTP protocol for media transport in
combination with the "RTP Profile for
Audio and Video Conferences with Minimal Control" over
UDP. The usage is explained in RFC 7826, .the same as RTP/AVP.Use of the RTP protocol for media transport in
combination with the "Extended RTP
Profile for RTCP-based Feedback (RTP/AVPF)" over UDP. The
usage is explained in RFC 7826, .the same as RTP/AVPF.Use of the RTP protocol for media transport in
combination with the "The Secure
Real-time Transport Protocol (SRTP)" over UDP. The usage
is explained in RFC 7826, .the same as RTP/SAVP.Use of the RTP
protocol for media transport in combination with the "Extended Secure RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)" over UDP. The usage is explained in RFC
7826, .the same as RTP/SAVPF.Use of the RTP protocol for media transport in
combination with the "RTP profile for
audio and video conferences with minimal control" over
TCP. The usage is explained in RFC 7826, .Use of the RTP protocol for media transport in
combination with the "Extended RTP
Profile for Real-time Transport Control Protocol (RTCP)-Based
Feedback (RTP/AVPF)" over TCP. The usage is explained in
RFC 7826, .Use of the RTP protocol for media transport in
combination with the "The Secure
Real-time Transport Protocol (SRTP)" over TCP. The usage
is explained in RFC 7826, .Use of the RTP protocol for media transport in
combination with the "Extended Secure RTP
Profile for Real-time Transport Control Protocol (RTCP)-Based
Feedback (RTP/SAVPF)" over TCP. The usage is explained in
RFC 7826, .The Transport Mode is a Transport
header parameter. It is used to identify the general mode of
media transport. The PLAY value registered defines a PLAYBACK mode,
where media flows from server to client.A registration for the transport parameter mode follows the Standards Action policy. The registration
request needs to meet the following requirements: A value definition that follows the ABNF "token" definition
.Text that explains how the registered value is used in
RTSP.This specification registers one value: See RFC 7826.The registry contains the transport mode value and the
reference.Transport Parameters are different parameters used in a Transport header's transport
specification to provide additional information required
beyond the transport protocol identifier to establish a functioning
transport.A registration for parameters that may be included in the
Transport header follows the Specification
Required policy. The expert reviewer verifies that the
registration request fulfills the following requirements:A Transport Parameter Name following the "token" ABNF
definition.A value definition, if the parameter takes a value, that
follows the ABNF definition of "trn-par-value" .Text that explains how the registered value is used in
RTSP. This specification registers all the transport parameters
defined in . This is a copy of that
list: unicastmulticastinterleavedttllayersssrcmodedest_addrsrc_addrsetupconnectionRTCP-muxMIKEYThe registry contains the transport parameter name and the
reference.This specification updates two URI schemes: one previously
registered, "rtsp", and one missing in the registry, "rtspu"
(previously only defined in RTSP 1.0).
One new URI scheme, "rtsps", is also registered. These URI schemes are
registered in an existing registry ("Uniform Resource Identifier (URI)
Schemes") not created by this memo. Registrations follow .rtspPermanentSee of RFC 7826.The rtsp scheme is used to
indicate resources accessible through the usage of the Real-Time
Streaming Protocol (RTSP). RTSP allows different operations on
the resource identified by the URI, but the primary purpose is
the streaming delivery of the resource to a client. However, the
operations that are currently defined are DESCRIBE,
GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
SETUP, SET_PARAMETER, and TEARDOWN.IRIs in this scheme are
defined and need to be encoded as RTSP URIs when used within
RTSP. That encoding is done according to RFC 3987.RTSP
1.0 (RFC 2326), RTSP 2.0 (RFC 7826).The extensions in
the URI syntax performed between RTSP 1.0 and 2.0 can create
interoperability issues. The changes are:Support for IPv6 literals in the host part and future IP
literals through a mechanism as defined in RFC 3986.A new relative format to use in RTSP elements that is not
required to start with "/".The above changes should have no impact on interoperability
as discussed in detail in of RFC
7826.All the security threats
identified in Section 7 of RFC 3986 also apply to this scheme.
They need to be reviewed and considered in any implementation
utilizing this scheme.Magnus Westerlund,
magnus.westerlund@ericsson.comIETFRFC 2326, RFC 3986, RFC 3987, and RFC
7826rtspsPermanentSee of RFC 7826.The rtsps scheme is used to
indicate resources accessible through the usage of the Real-Time
Streaming Protocol (RTSP) over TLS. RTSP allows different
operations on the resource identified by the URI, but the
primary purpose is the streaming delivery of the resource to a
client. However, the operations that are currently defined are
DESCRIBE, GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE,
REDIRECT, SETUP, SET_PARAMETER, and TEARDOWN.IRIs in this scheme are
defined and need to be encoded as RTSP URIs when used within
RTSP. That encoding is done according to RFC 3987.RTSP
1.0 (RFC 2326), RTSP 2.0 (RFC 7826).The "rtsps"
scheme was never officially defined for RTSP 1.0; however, it
has seen widespread use in actual deployments of RTSP 1.0.
Therefore, this section discusses the believed changes between
the unspecified RTSP 1.0 "rtsps" scheme and RTSP 2.0 definition.
The extensions in the URI syntax performed between RTSP 1.0 and
2.0 can create interoperability issues. The changes are:Support for IPv6 literals in the host part and future IP
literals through a mechanism as defined by RFC 3986.A new relative format to use in RTSP elements that is not
required to start with "/".The above changes should have no impact on interoperability
as discussed in detail in of RFC
7826.All the security threats
identified in Section 7 of RFC 3986 also apply to this scheme.
They need to be reviewed and considered in any implementation
utilizing this scheme.Magnus Westerlund,
magnus.westerlund@ericsson.comIETFRFC 2326, RFC 3986, RFC 3987, and RFC
7826rtspuPermanentSee Section 3.2 of RFC
2326.The rtspu scheme is used to
indicate resources accessible through the usage of the Real-Time
Streaming Protocol (RTSP) over unreliable datagram transport.
RTSP allows different operations on the resource identified by
the URI, but the primary purpose is the streaming delivery of
the resource to a client. However, the operations that are
currently defined are DESCRIBE, GET_PARAMETER, OPTIONS,
REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and
TEARDOWN.This scheme is not
intended to be used with characters outside the US-ASCII
repertoire.RTSP
1.0 (RFC 2326).The definition of
the transport mechanism of RTSP over UDP has interoperability
issues. That makes the usage of this scheme problematic.All the security threats
identified in Section 7 of RFC 3986 also apply to this scheme.
They need to be reviewed and considered in any implementation
utilizing this scheme.Magnus Westerlund,
magnus.westerlund@ericsson.comIETFRFC 2326This specification defines three SDP
attributes that have been registered by IANA.textparameterscharset: The charset parameter
is applicable to the encoding of the parameter values. The default
charset is UTF&nbhy;8, if the 'charset' parameter is not
present.8bitThis format may carry any
type of parameters. Some can have security requirements, like
privacy, confidentiality, or integrity requirements. The format
has no built-in security protection. For the usage, the transport
can be protected between server and client using TLS. However,
care must be taken to consider if the proxies are also trusted
with the parameters in case hop-by-hop security is used. If stored
as a file in a file system, the necessary precautions need to be
taken in relation to the parameter requirements including object
security such as S/MIME .This media type was
mentioned as a fictional example in , but
was not formally specified. This has resulted in usage of this
media type that may not match its formal definition.RFC 7826, .Applications
that use RTSP and have additional parameters they like to read and
set using the RTSP GET_PARAMETER and SET_PARAMETER methods.N/AN/AN/A
Magnus Westerlund (magnus.westerlund@ericsson.com)CommonNoneMagnus Westerlund
(magnus.westerlund@ericsson.com)IETFTransparent end-to-end Packet-switched Streaming Service
(PSS); Protocols and codecs3rd Generation Partnership Project
(3GPP)Federal Information Processing Standards Publication: Secure
Hash Standard (SHS)National Institute of Standards and Technology
(NIST)A Network Address Translator (NAT) Traversal Mechanism for
Media Controlled by Real-Time Streaming Protocol (RTSP)ST 12-1:2008 For Television -- Time and Control CodeSociety of Motion Picture and Television
EngineersMultimedia Congestion Control: Circuit Breakers for Unicast
RTP SessionsInformation technology -- Generic coding of moving pictures
and associated audio information - part 6: Extension for
DSM-CCInternational Organization for
StandardizationData elements and interchange formats - Information
interchange - Representation of dates and timesInternational Organization for
StandardizationUnix Networking Programming, Volume 1: The Sockets Networking
API (3rd Edition)This section contains several different examples trying to illustrate
possible ways of using RTSP. The examples can also help with the
understanding of how functions of RTSP work. However, remember that
these are examples and the normative and syntax descriptions in the
other sections take precedence. Please also note that many of the
examples have been broken into several lines, where following lines
start with whitespace as allowed by the syntax.This is an example of media-on-demand streaming of media stored in
a container file. For the purposes of this example, a container file
is a storage entity in which multiple continuous media types
pertaining to the same end-user presentation are present. In effect,
the container file represents an RTSP presentation, with each of its
components being RTSP-controlled media streams. Container files are a
widely used means to store such presentations. While the components
are transported as independent streams, it is desirable to maintain a
common context for those streams at the server end.This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server.It is also possible that the presentation author may wish to
prevent selective retrieval of the streams by the client in order to
preserve the artistic effect of the combined media presentation.
Similarly, in such a tightly bound presentation, it is desirable to be
able to control all the streams via a single control message using an
aggregate URI.The following is an example of using a single RTSP session to
control multiple streams. It also illustrates the use of aggregate
URIs. In a container file, it is also desirable not to write any URI
parts that are not kept when the container is distributed, like the
host and most of the path element. Therefore, this example also uses
the "*" and relative URI in the delivered SDP.Also, this presentation description (SDP) is not cacheable, as the
Expires header is set to an equal value with date indicating immediate
expiration of its validity.Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to the
container file.This example is basically the example above (), but now utilizing pipelining to
speed up the setup. It requires only two round-trip times until the
media starts flowing. First of all, the session description is
retrieved to determine what media resources need to be set up. In the
second step, one sends the necessary SETUP requests and the PLAY
request to initiate media delivery.Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to the
container file.This example is basically the above example (), but now including
establishment of SRTP crypto contexts to get a secured media delivery.
First of all, the client attempts to fetch this insecurely, but the
server redirects to a URI indicating a requirement on using a secure
connection for the RTSP messages. The client establishes a TCP/TLS
connection, and the session description is retrieved to determine what
media resources need to be set up. In the this session description,
secure media (SRTP) is indicated. In the next step, the client sends
the necessary SETUP requests including MIKEY messages. This is
pipelined with a PLAY request to initiate media delivery.Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to the
container file.Note: The MIKEY messages below are not valid MIKEY messages and are
Base64-encoded random data to represent where the MIKEY messages would
go.An alternative example of media on demand with a few more tweaks is
the following. Client C requests a movie distributed from two
different media servers A (audio.example.com) and V
(video.example.com). The media description is stored on a web server
W. The media description contains descriptions of the presentation and
all its streams, including the codecs that are available and the
protocol stack.In this example, the client is only interested in the last part of
the movie.Even though the audio and video track are on two different servers
that may start at slightly different times and may drift with respect
to each other over time, the client can perform initial
synchronization of the two media using RTP-Info and Range received in
the PLAY responses. If the two servers are time synchronized, the RTCP
packets can also be used to maintain synchronization.Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients needs to use the rules set forth in the session
description for Request-URIs rather than assuming that a consistent
URI may always be used throughout. Below is an example of how a
multi-stream server might expect a single-stream file to be served:
Note the different URI in the SETUP command and then the switch
back to the aggregate URI in the PLAY command. This makes complete
sense when there are multiple streams with aggregate control, but it
is less than intuitive in the special case where the number of streams
is one. However, the server has declared the aggregated control URI in
the SDP; therefore, this is legal.In this case, it is also required that servers accept
implementations that use the non-aggregated interpretation and use the
individual media URI, like this:The media server M chooses the multicast address and port. Here, it
is assumed that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
This example illustrates how the client and server determine their
capability to support a special feature, in this case, "play.scale".
The server, through the client request and the included Supported
header, learns that the client supports RTSP 2.0 and also supports the
playback time scaling feature of RTSP. The server's response contains
the following feature-related information to the client; it supports
the basic media delivery functions (play.basic), the extended
functionality of time scaling of content (play.scale), and one
"example.com" proprietary feature (com.example.flight). The client
also learns the methods supported (Public header) by the server for
the indicated resource.When the client sends its SETUP request, it tells the server that
it requires support of the play.scale feature for this session by
including the Require header.The RTSP session state machine describes the behavior of the protocol
from RTSP session initialization through RTSP session termination. It is
probably easiest to think of this as the server's state and then view
the client as needing to track what it believes the server's state will
be based on sent or received RTSP messages. Thus, in most cases, the
state tables below can be read as: if the client does X, and assuming it
fulfills any prerequisite(s), the (server) state will move to the new
state and the indicated response will returned. However, there are also
server-to-client notifications or requests, where the action describes
what notification or request will occur, its requisites, what new state
will result after the server has received the response, as well as
describing the client's response to the action.The State machine is defined on a per-session basis, which is
uniquely identified by the RTSP session identifier. The session may
contain one or more media streams depending on state. If a single media
stream is part of the session, it is in non-aggregated control. If two
or more are part of the session, it is in aggregated control.The below state machine is an informative description of the
protocol's behavior. In case of ambiguity with the earlier parts of this
specification, the description in the earlier parts take precedence.The state machine contains three states, described below. For each
state, there exists a table that shows which requests and events are
allowed and whether they will result in a state change. Initial state, no session exists.Session is ready to start playing.Session is playing, i.e., sending media-stream
data in the direction S->C.This representation of the state machine needs more than its state
to work. A small number of variables are also needed, and they are
explained below. The number of media streams that are part of
this session.Resume point, the point in the presentation time
line at which a request to continue playing will resume from. A
time format for the variable is not mandated.To make the state tables more compact, a number of abbreviations
are used, which are explained below. IF Implemented.MediaPause Point, the point in the presentation
timeline at which the presentation was paused.Presentation, the complete multimedia
presentation.Redirect Point, the point in the presentation
timeline at which a REDIRECT was specified to occur.Session.This section contains a table for each state. The table contains
all the requests and events on which this state is allowed to act. The
events that are method names are, unless noted, requests with the
given method in the direction client to server (C->S). In some
cases, there exists one or more requisites. The response column tells
what type of response actions should be performed. Possible actions
that are requested for an event include: response codes, e.g., 200,
headers that need to be included in the response, setting of state
variables, or settings of other session-related parameters. The new
state column tells which state the state machine changes to.The response to a valid request meeting the requisites is normally
a 2xx (SUCCESS) unless otherwise noted in the response column. The
exceptions need to be given a response according to the response
column. If the request does not meet the requisite, is erroneous, or
some other type of error occurs, the appropriate response code is to
be sent. If the response code is a 4xx, the session state is
unchanged. A response code of 3rr will result in that the session
being ended and its state changed to Init. A response code of 304
results in no state change. However, there are restrictions to when a
3rr response may be used. A 5xx response does not result in any change
of the session state, except if the error is not possible to recover
from. An unrecoverable error results in the ending of the session. In
the general case, if it can't be determined whether or not it was an
unrecoverable error, the client will be required to test. In the case
that the next request after a 5xx is responded to with a 454 (Session
Not Found), the client knows that the session has ended. For any
request message that cannot be responded to within the time defined in
, a 100 response must be
sent.The server will time out the session after the period of time
specified in the SETUP response, if no activity from the client is
detected. Therefore, there exists a timeout event for all states
except Init.In the case that NRM = 1, the presentation URI is equal to the
media URI or a specified presentation URI. For NRM > 1, the
presentation URI needs to be other than any of the media that are part
of the session. This applies to all states.EventPrerequisiteResponseDESCRIBENeeds REDIRECT3rr, RedirectDESCRIBE200, Session descriptionOPTIONSSession ID200, Reset session timeout timerOPTIONS200SET_PARAMETERValid parameter200, change value of parameterGET_PARAMETERValid parameter200, return value of parameterThe methods in do not have any
effect on the state machine or the state variables. However, some
methods do change other session-related parameters, for example,
SET_PARAMETER, which will set the parameter(s) specified in its body.
Also, all of these methods that allow the Session header will also
update the keep-alive timer for the session.ActionRequisiteNew StateResponseSETUPReadyNRM=1, RP=0.0SETUPNeeds RedirectInit3rr RedirectS -> C: REDIRECTNo Session hdrInitTerminate all SESThe initial state of the state machine () can only be left by processing a correct
SETUP request. As seen in the table, the two state variables are also
set by a correct request. This table also shows that a correct SETUP
can in some cases be redirected to another URI or server by a 3rr
response.ActionRequisiteNew StateResponseSETUPNew URIReadyNRM +=1SETUPURI Setup priorReadyChange transport paramTEARDOWNPrs URI,InitNo session hdr, NRM = 0TEARDOWNmd URI,NRM=1InitNo Session hdr, NRM = 0TEARDOWNmd URI,NRM>1ReadySession hdr, NRM -= 1PLAYPrs URI, No rangePlayPlay from RPPLAYPrs URI, RangePlayAccording to rangePLAYmd URI, NRM=1, RangePlayAccording to rangePLAYmd URI, NRM=1PlayPlay from RPPAUSEPrs URIReadyReturn PPSC:REDIRECTTerminate-ReasonReadySet RedPSC:REDIRECTNo Terminate-Reason time parameterInitSession is removedTimeoutInitRedP reachedInitTEARDOWN of sessionIn the Ready state (), some of the
actions depend on the number of media streams (NRM) in the session,
i.e., aggregated or non-aggregated control. A SETUP request in the
Ready state can either add one more media stream to the session or, if
the media stream (same URI) already is part of the session, change the
transport parameters. TEARDOWN depends on both the Request-URI and the
number of media streams within the session. If the Request-URI is the
presentation URI, the whole session is torn down. If a media URI is
used in the TEARDOWN request and more than one media exists in the
session, the session will remain and a session header is returned in
the response. If only a single media stream remains in the session
when performing a TEARDOWN with a media URI, the session is removed.
The number of media streams remaining after tearing down a media
stream determines the new state.ActionRequisiteNew StateResponsePAUSEPrs URIReadySet RP to present pointEnd of mediaAll mediaPlaySet RP = End of mediaEnd of rangePlaySet RP = End of rangePLAYPrs URI, No rangePlayPlay from present pointPLAYPrs URI, RangePlayAccording to rangeSC:PLAY_NOTIFYPlay200SETUPNew URIPlay455SETUPmd URIPlay455SETUPmd URI, IFIPlayChange transport param.TEARDOWNPrs URIInitNo session hdrTEARDOWNmd URI,NRM=1InitNo Session hdr, NRM=0TEARDOWNmd URIPlay455SC:REDIRECTTerminate Reason with Time parameterPlaySet RedPSC:REDIRECTInitSession is removedRedP reachedInitTEARDOWN of sessionTimeoutInitStop Media playoutThe Play state table () contains a
number of requests that need a presentation URI (labeled as Prs URI)
to work on (i.e., the presentation URI has to be used as the
Request-URI). This is due to the exclusion of non-aggregated stream
control in sessions with more than one media stream.To avoid inconsistencies between the client and server, automatic
state transitions are avoided. This can be seen at, for example, an
"End of media" event when all media has finished playing but the
session still remains in Play state. An explicit PAUSE request needs
to be sent to change the state to Ready. It may appear that there
exist automatic transitions in "RedP reached" and "PP reached".
However, they are requested and acknowledged before they take place.
The time at which the transition will happen is known by looking at
the Terminate-Reason header's time parameter and Range header,
respectively. If the client sends a request close in time to these
transitions, it needs to be prepared for receiving error messages, as
the state may or may not have changed.This section defines how certain combinations of protocols, profiles,
and lower transports are used. This includes the usage of the Transport
header's source and destination address parameters: "src_addr" and
"dest_addr".This section defines the interaction of RTSP with respect to the
RTP protocol . It also defines any necessary
media-transport signaling with regard to RTP.The available RTP profiles and lower-layer transports are described
below along with rules on signaling the available combinations.The usage of the "RTP Profile for Audio and Video Conferences
with Minimal Control" when using RTP for
media transport over different lower-layer transport protocols is
defined below in regard to RTSP.One such case is defined within this document: the use of
embedded (interleaved) binary data as defined in . The usage of this method is indicated by
including the "interleaved" parameter.When using embedded binary data, "src_addr" and "dest_addr" MUST
NOT be used. This addressing and multiplexing is used as defined
with use of channel numbers and the interleaved parameter.This part describes the sending of RTP
over lower-transport-layer UDP according to
the profile "RTP Profile for Audio and Video Conferences with
Minimal Control" defined in .
Implementations of RTP/AVP/UDP MUST implement RTCP. This profile requires
one or two unidirectional or bidirectional UDP flows per media
stream. The first UDP flow is for RTP and the second is for RTCP.
Multiplexing of RTP and RTCP
MAY be used, in which case, a single UDP flow is used for both
parts. Embedding of RTP data with the RTSP messages, in accordance
with , SHOULD NOT be performed when RTSP
messages are transported over unreliable transport protocols, like
UDP .The RTP/UDP and RTCP/UDP flows can be established using the
Transport header's "src_addr" and "dest_addr" parameters.In RTSP PLAY mode, the transmission of RTP packets from client to
server is unspecified. The behavior in regard to such RTP packets
MAY be defined in future.The "src_addr" and "dest_addr" parameters are used in the
following way for media delivery and playback mode, i.e., Mode=PLAY:
The "src_addr" and "dest_addr" parameters MUST contain either
1 or 2 address specifications. Note that two address
specifications MAY be provided even if RTP and RTCP multiplexing
is negotiated.Each address specification for RTP/AVP/UDP or RTP/AVP/TCP
MUST contain either: both an address and a port number, ora port number without an address.The first address specification given in either of the
parameters applies to the RTP stream. The second specification,
if present, applies to the RTCP stream, unless in the case RTP
and RTCP multiplexing is negotiated where both RTP and RTCP will
use the first specification.The RTP/UDP packets from the server to the client MUST be
sent to the address and port given by the first address
specification of the "dest_addr" parameter.The RTCP/UDP packets from the server to the client MUST be
sent to the address and port given by the second address
specification of the "dest_addr" parameter, unless RTP and RTCP
multiplexing has been negotiated, in which case RTCP MUST be
sent to the first address specification. If no second pair is
specified and RTP and RTCP multiplexing has not been negotiated,
RTCP MUST NOT be sent.The RTCP/UDP packets from the client to the server MUST be
sent to the address and port given by the second address
specification of the "src_addr" parameter, unless RTP and RTCP
multiplexing has been negotiated, in which case RTCP MUST be
sent to the first address specification. If no second pair is
specified and RTP and RTCP multiplexing has not been negotiated,
RTCP MUST NOT be sent.The RTP/UDP packets from the client to the server MUST be
sent to the address and port given by the first address
specification of the "src_addr" parameter.RTP and RTCP packets SHOULD be sent from the corresponding
receiver port, i.e., RTCP packets from the server should be sent
from the "src_addr" parameters second address port pair, unless
RTP and RTCP multiplexing has been negotiated in which case the
first address port pair is used.The RTP profile "Extended RTP Profile for
RTCP-based Feedback (RTP/AVPF)" MAY be used as RTP profiles
in sessions using RTP. All that is defined for AVP MUST also apply
for AVPF.The usage of AVPF is indicated by the media initialization
protocol used. In the case of SDP, it is indicated by media lines
("m=") containing the profile RTP/AVPF. That SDP MAY also contain
further AVPF-related SDP attributes configuring the AVPF session
regarding reporting interval and feedback messages to be used . This configuration MUST be followed.The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
is an RTP profile (SAVP) that MAY be used
in RTSP sessions using RTP. All that is defined for AVP MUST also
apply for SAVP.The usage of SRTP requires that a security context be
established. The default key-management unless otherwise signaled
SHALL be MIKEY in RSA-R mode as defined in and not according to the procedure defined in
"Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming Protocol
(RTSP)". The reason is that RFC 4567 sends the initial MIKEY
message in SDP, thus, both requiring the usage of the DESCRIBE
method and forcing the server to keep state for clients performing
DESCRIBE in anticipation that they might require key management.MIKEY is selected as the default method for establishing SRTP
cryptographic context within an RTSP session as it can be embedded
in the RTSP messages while still ensuring confidentiality of content
of the keying material, even when using hop-by-hop TLS security for
the RTSP messages. This method also supports pipelining of the RTSP
messages.This method for using MIKEY to
establish the SRTP cryptographic context is initiated in the
client's SETUP request, and the server's response to the SETUP
carries the MIKEY response. This ensures that the crypto context
establishment happens simultaneously with the establishment of the
media stream being protected. By using MIKEY's RSA-R mode the client can be the initiator
and still allow the server to set the parameters in accordance
with the actual media stream.The SRTP cryptographic context establishment is done according
to the following process:The client determines that SAVP or SAVPF shall be used from
the media-description format, e.g., SDP. If no other
key-management method is explicitly signaled, then MIKEY SHALL
be used as defined herein. The use of SRTP with RTSP is only
defined with MIKEY with keys established as defined in this
section. Future documents may define how an RTSP
implementation treats SDP that indicates some other key
mechanism to be used. The need for such specification includes
, which is not defined for use in RTSP
2.0 within this document.The client SHALL establish a TLS connection for RTSP
messages, directly or hop-by-hop with the server. If
hop-by-hop TLS security is used, the User method SHALL be
indicated in the Accept-Credentials header. Note that using
hop-by-hop does allow the proxy to insert itself as a man in
the middle. This can also occur in the MIKEY exchange by the
proxy providing one of its certificates rather than the
server's in the Connection-Credentials header. Therefore, the
client SHALL validate the server certificate.The client retrieves the server's certificate from a direct
TLS connection or hop-by-hop from a Connection-Credentials
header. The client then checks that the server certificate is
valid and belongs to the server.The client forms the MIKEY Initiator message using RSA-R
mode in unicast mode as specified in .
The client SHOULD use the same certificate for TLS and MIKEY
to enable the server to bind the two together. The client's
certificate SHALL be included in the MIKEY message. The client
SHALL indicate its SRTP capabilities in the message.The MIKEY message from the previous step is base64-encoded and becomes the value
of the MIKEY parameter that is included in the transport
specification(s) that specifies an SRTP-based profile (SAVP,
SAVPF) in the SETUP request.Any proxy encountering the MIKEY parameter SHALL forward it
without modification. A proxy that is required to understand
the Transport specifications will need to understand
SAVP/SAVPF with MIKEY to enable the default keying for
SRTP-protected media streams. If such a proxy does not support
SAVP/SAVPF with MIKEY, it will discard the whole transport
specification. Most types of proxies can easily support SAVP
and SAVPF with MIKEY. If a client encounters a proxy not
supporting SAVP/SAVPF with MIKEY, the client should attempt
bypassing that proxy.The server, upon receiving the SETUP request, will need to
decide upon the transport specification to use, if multiple
are included by the client. In the determination of which
transport specifications are supported and preferred, the
server SHOULD decode the MIKEY message to take the embedded
SRTP parameters into account. If all transport spec require
SRTP but no MIKEY parameter or other supported keying method
is included, the server SHALL respond with 403
(Forbidden).Upon generating a response, the following outcomes can
occur:A transport spec not using SRTP and MIKEY is selected.
Thus, the response will not contain any MIKEY
parameters.A transport spec using SRTP and MIKEY is selected but
an error is encountered in the MIKEY processing. In this
case, an RTSP error response code of 466 (Key Management
Error) SHALL be used. A MIKEY message describing the error
MAY be included.A transport spec using SRTP and MIKEY is selected and a
MIKEY response message can be created. The server SHOULD
use the same certificate for TLS and in MIKEY to enable
the client to bind the two together. If a different
certificate is used, it SHALL be included in the MIKEY
message. It is RECOMMENDED that the envelope key-cache
type be set to ‘Cache’ and that a single
envelope key is reused for all MIKEY messages to the
client. That message is included in the MIKEY parameter
part of the single selected transport specification in the
SETUP response. The server will set the SRTP parameters as
preferred for this media stream within the supported range
by the client.The server transmits the SETUP response back to the
client.The client receives the SETUP response and, if the response
code indicates a successful request, it decodes the MIKEY
message and establishes the SRTP cryptographic context from
the parameters in the MIKEY response.In the above method, the client's certificate may be self
signed in cases where the client's identity is not necessary to
authenticate and the security goal is only to ensure that the RTSP
signaling client is the same as the one receiving the SRTP
security context.The RTP profile "Extended Secure RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" is an RTP profile (SAVPF) that MAY be used in
RTSP sessions using RTP. All that is defined for AVPF MUST also
apply for SAVPF.The usage of SRTP requires that a cryptographic context be
established. The default mechanism for establishing that security
association is to use MIKEY with RTSP as
defined in .RTCP has several usages when RTP is implemented for media
transport as explained below. Thus, RTCP MUST be supported if an
RTSP agent handles RTP.RTCP provides media synchronization and clock-drift
compensation. The initial media synchronization is available from
RTP-Info header. However, to be able to handle any clock drift
between the media streams, RTCP is needed.RTCP traffic from the RTSP client to the RTSP server MUST
function as keep-alive. This requires an RTSP server supporting
RTP to use the received RTCP packets as indications that the
client desires the related RTSP session to be kept alive.RTCP Receiver reports and any additional feedback from the
client MUST be used to adapt the bitrate used over the transport
for all cases when RTP is sent over UDP. An RTP sender without
reserved resources MUST NOT use more than its fair share of the
available resources. This can be determined by comparing on
short-to-medium terms (some seconds) the used bitrate and adapting
it so that the RTP sender sends at a bitrate comparable to what a
TCP sender would achieve on average over the same path.To ensure that the implementation's adaptation mechanism has a
well-defined outer envelope, all implementations using a
non-congestion-controlled unicast transport protocol, like UDP,
MUST implement "Multimedia
Congestion Control: Circuit Breakers for Unicast RTP
Sessions".RTSP can be used to negotiate the usage of RTP and RTCP
multiplexing as described in . This allows
servers and client to reduce the amount of resources required for
the session by only requiring one underlying transport stream per
media stream instead of two when using RTP and RTCP. This lessens
the server-port consumption and also the necessary state and
keep-alive work when operating across NATs.Content must be prepared with some consideration for RTP and
RTCP multiplexing, mainly ensuring that the RTP payload types used
do not collide with the ones used for RTCP packet types. This
option likely needs explicit support from the content unless the
RTP payload types can be remapped by the server and that is
correctly reflected in the session description. Beyond that,
support of this feature should come at little cost and much
gain.It is recommended that, if the content and server support RTP
and RTCP multiplexing, this is indicated in the session
description, for example, using the SDP attribute "a=rtcp-mux". If
the SDP message contains the "a=rtcp-mux" attribute for a media
stream, the server MUST support RTP and RTCP multiplexing. If
indicated or otherwise desired by the client, it can include the
Transport parameter "RTCP-mux" in any transport specification
where it desires to use "RTCP-mux". The server will indicate if it
supports "RTCP-mux". Servers and Clients SHOULD support RTP and
RTCP multiplexing.For capability exchange, an RTSP feature tag for RTP and RTCP
multiplexing is defined: "setup.rtp.rtcp.mux".To minimize the risk of negotiation failure while using RTP and
RTCP multiplexing, some recommendations are here provided. If the
session description includes explicit indication of support
("a=rtcp-mux" in SDP), then an RTSP agent can safely create a
SETUP request with a transport specification with only a single
"dest_addr" parameter address specification. If no such explicit
indication is provided, then even if the feature tag
"setup.rtp.rtcp.mux" is provided in a Supported header by the RTSP
server or the feature tag included in the Required header in the
SETUP request, the media resource may not support RTP and RTCP
multiplexing. Thus, to maximize the probability of successful
negotiation, the RTSP agent is recommended to include two
"dest_addr" parameter address specifications in the first or first
set (if pipelining is used) of SETUP request(s) for any media
resource aggregate. That way, the RTSP server can accept RTP and
RTCP multiplexing and only use the first address specification or,
if not, use both specifications. The RTSP agent, after having
received the response for a successful negotiation of the usage of
RTP and RTCP multiplexing, can then release the resources
associated with the second address specification.Transport of RTP over TCP can be done in two ways: over independent
TCP connections using or interleaved in the
RTSP connection. In both cases, the protocol MUST be "rtp" and the
lower-layer MUST be TCP. The profile may be any of the above specified
ones: AVP, AVPF, SAVP, or SAVPF.The use of embedded (interleaved) binary data transported on the
RTSP connection is possible as specified in . When using this declared combination of
interleaved binary data, the RTSP messages MUST be transported over
TCP. TLS may or may not be used. If TLS is used, both RTSP messages
and the binary data will be protected by TLS.One should, however, consider that this will result in all media
streams going through any proxy. Using independent TCP connections
can avoid that issue.In this section, the sending of RTP over
lower-layer transport TCP according to
"Framing Real-time Transport Protocol (RTP) and RTP Control Protocol
(RTCP) Packets over Connection-Oriented Transport" is described. This section adapts the guidelines
for using RTP over TCP within SIP/SDP to
work with RTSP.A client codes the support of RTP over independent TCP by
specifying an RTP/AVP/TCP transport option without an interleaved
parameter in the Transport line of a SETUP request. This transport
option MUST include the "unicast" parameter.If the client wishes to use RTP with RTCP, two address
specifications need to be included in the "dest_addr" parameter. If
the client wishes to use RTP without RTCP, one address specification
is included in the "dest_addr" parameter. If the client wishes to
multiplex RTP and RTCP on a single transport flow (see ), one or two address specifications are
included in the "dest_addr" parameter in addition to the "RTCP-mux"
transport parameter. Two address specifications are allowed to
facilitate successful negotiation when the server or content can't
support RTP and RTCP multiplexing. Ordering rules of dest_addr ports
follow the rules for RTP/AVP/UDP.If the client wishes to play the active role in initiating the
TCP connection, it MAY set the setup parameter (see ) on the Transport line to be "active", or
it MAY omit the setup parameter, as active is the default. If the
client signals the active role, the ports in the address
specifications in the "dest_addr" parameter MUST be set to 9 (the
discard port).If the client wishes to play the passive role in TCP connection
initiation, it MUST set the setup parameter on the Transport line to
be "passive". If the client is able to assume the active or the
passive role, it MUST set the setup parameter on the Transport line
to be "actpass". In either case, the "dest_addr" parameter's address
specification port value for RTP MUST be set to the TCP port number
on which the client is expecting to receive the TCP connection for
RTP, and the "dest_addr" address specification port value for RTCP
MUST be set to the TCP port number on which the client is expecting
to receive the TCP connection for RTCP. In the case that the client
wishes to multiplex RTP and RTCP on a single transport flow, the
"RTCP-mux" parameter is included and one or two "dest_addr"
parameter address specifications are included, as mentioned earlier
in this section.Upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, if a
server decides to accept this requested option, the 2xx reply MUST
contain a Transport option that specifies RTP/AVP/TCP (without using
the interleaved parameter and using the unicast parameter). The
"dest_addr" parameter value MUST be echoed from the parameter value
in the client request unless the destination address (only port) was
not provided; in which case, the server MAY include the source
address of the RTSP TCP connection with the port number
unchanged.In addition, the server reply MUST set the setup parameter on the
Transport line, to indicate the role the server will play in the
connection setup. Permissible values are "active" (if a client set
setup to "passive" or "actpass") and "passive" (if a client set
setup to "active" or "actpass").If a server sets setup to "passive", the "src_addr" in the reply
MUST indicate the ports on which the server is willing to receive a
TCP connection for RTP and (if the client requested a TCP connection
for RTCP by specifying two "dest_addr" address specifications) a
TCP/RTCP connection. If a server sets setup to "active", the ports
specified in "src_addr" address specifications MUST be set to 9. The
server MAY use the "ssrc" parameter, following the guidance in . The server sets only one address
specification in the case that the client has indicated only a
single address specification or in case RTP and RTCP multiplexing
was requested and accepted by the server. Port ordering for
"src_addr" follows the rules for RTP/AVP/UDP.Servers MUST support taking the passive role and MAY support
taking the active role. Servers with a public IP address take the
passive role, thus enabling clients behind NATs and firewalls a
better chance of successful connect to the server by actively
connecting outwards. Therefore, the clients are RECOMMENDED to take
the active role.After sending (receiving) a 2xx reply for a SETUP method for a
non-interleaved RTP/AVP/TCP media stream, the active party SHOULD
initiate the TCP connection as soon as possible. The client MUST NOT
send a PLAY request prior to the establishment of all the TCP
connections negotiated using SETUP for the session. In case the
server receives a PLAY request in a session that has not yet
established all the TCP connections, it MUST respond using the 464
(Data Transport Not Ready Yet) () error
code.Once the PLAY request for a media resource transported over
non-interleaved RTP/AVP/TCP occurs, media begins to flow from server
to client over the RTP TCP connection, and RTCP packets flow
bidirectionally over the RTCP TCP connection. Unless RTP and RTCP
multiplexing has been negotiated; in which case, RTP and RTCP will
flow over a common TCP connection. As in the RTP/UDP case,
client-to-server traffic on an RTP-only TCP session is unspecified
by this memo. The packets that travel on these connections MUST be
framed using the protocol defined in , not
by the framing defined for interleaving RTP over the RTSP connection
defined in .A successful PAUSE request for media being transported over
RTP/AVP/TCP pauses the flow of packets over the connections, without
closing the connections. A successful TEARDOWN request signals that
the TCP connections for RTP and RTCP are to be closed by the RTSP
client as soon as possible.Subsequent SETUP requests using a URI already set up in an RTSP
session using an RTP/AVP/TCP transport specification may be
ambiguous in the following way: does the client wish to open up a
new TCP connection for RTP or RTCP for the URI, or does the client
wish to continue using the existing TCP connections? The client
SHOULD use the "connection" parameter (defined in ) on the Transport line to make its
intention clear (by setting "connection" to "new" if new connections
are needed, and by setting "connection" to "existing" if the
existing connections are to be used). After a 2xx reply for a SETUP
request for a new connection, parties should close the preexisting
connections, after waiting a suitable period for any stray RTP or
RTCP packets to arrive.The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires
that a security association be established. The default mechanism
for establishing that security association is to use MIKEY with RTSP as defined .Below, a rewritten version of the example "Media on Demand" shows the
use of RTP/AVP/TCP non-interleaved:RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP media layer. Two cases occur, the first is
when a new PLAY request replaces an old ongoing request and the new
request results in a jump in the media. This should produce continuous
media stream at the RTP layer. A client may also immediately follow a
completed PLAY request with a new PLAY request. This will result in
some gap in the media layer. The below text will look into both
cases.A PLAY request that replaces an ongoing PLAY request allows the
media layer rendering the RTP stream to do so continuously without
being affected by jumps in media-clock time. The RTP timestamps for
the new media range are set so that they become continuous with the
previous media range in the previous request. The RTP sequence number
for the first packet in the new range will be the next following the
last packet in the previous range, i.e., monotonically increasing. The
goal is to allow the media-rendering layer to work without
interruption or reconfiguration across the jumps in media clock. This
should be possible in all cases of replaced PLAY requests for media
that has random access properties. In this case, care is needed to
align frames or similar media-dependent structures.In cases where jumps in media-clock time are a result of RTSP
signaling operations arriving after a completed PLAY operation, the
request timing will result in that media becoming non-continuous. The
server becomes unable to send the media so that it arrives timely and
still carries timestamps to make the media stream continuous. In these
situations, the server will produce RTP streams where there are gaps
in the RTP timeline for the media. If the media has frame structure,
aligning the timestamp for the next frame with the previous structure
reduces the burden to render this media. The gap should represent the
time the server hasn't been serving media, e.g., the time between the
end of the media stream or a PAUSE request and the new PLAY request.
In these cases, the RTP sequence number would normally be
monotonically increasing across the gap.For RTSP sessions with media that lacks random access properties,
such as live streams, any media-clock jump is commonly the result of a
correspondingly long pause of delivery. The RTP timestamp will have
increased in direct proportion to the duration of the paused delivery.
Note also that in this case the RTP sequence number should be the next
packet number. If not, the RTCP packet loss reporting will indicate as
loss all packets not received between the point of pausing and later
resuming. This may trigger congestion avoidance mechanisms. An allowed
exception from the above recommendation on monotonically increasing
RTP sequence number is live media streams, likely being relayed. In
this case, when the client resumes delivery, it will get the media
that is currently being delivered to the server itself. For this type
of basic delivery of live streams to multiple users over unicast,
individual rewriting of RTP sequence numbers becomes quite a burden.
For solutions that already cache media or perform time shifting, the
rewriting should impose only a minor burden.The goal when handling jumps in media-clock time is that the
provided stream is continuous without gaps in RTP timestamp or
sequence number. However, when delivery has been halted for some
reason, the RTP timestamp, when resuming, MUST represent the duration
that the delivery was halted. An RTP sequence number MUST generally be
the next number, i.e., monotonically increasing modulo 65536. For
media resources with the properties Time-Progressing and
Time-Duration=0.0, the server MAY create RTP media streams with RTP
sequence number jumps in them due to the client first halting delivery
and later resuming it (PAUSE and then later PLAY). However, servers
utilizing this exception must take into consideration the resulting
RTCP receiver reports that likely contain loss reports for all the
packets that were a part of the discontinuity. A client cannot rely on
the fact that a server will align when resuming play, even if it is
RECOMMENDED. The RTP-Info header will provide information on how the
server acts in each case.One cannot assume that the RTSP client can communicate with the
RTP media agent, as the two may be independent processes. If the
RTP timestamp shows the same gap as the NPT, the media agent will
assume that there is a pause in the presentation. If the jump in
NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just
played out. Having the RTP timestamp jump will also affect the
RTCP measurements based on this.As an example, assume an RTP timestamp frequency of 8000 Hz, a
packetization interval of 100 ms, and an initial sequence number and
timestamp of zero.The ensuing RTP data stream is depicted below:Upon the completion of the requested delivery, the server sends a
PLAY_NOTIFY.Upon the completion of the play range, the client follows up with a
request to PLAY from a new NPT.The ensuing RTP data stream is depicted below:In this example, first, NPT 10 through 15 are played, then the
client requests the server to skip ahead and play NPT 18 through 20.
The first segment is presented as RTP packets with sequence numbers 0
through 49 and timestamps 0 through 39,200. The second segment
consists of RTP packets with sequence numbers 50 through 69, with
timestamps 40,100 through 55,200. While there is a gap in the NPT,
there is no gap in the sequence-number space of the RTP data
stream.The RTP timestamp gap is present in the above example due to the
time it takes to perform the second play request, in this case, 12.5
ms (100/8000).During a PAUSE/PLAY interaction in an RTSP session, the duration of
time for which the RTP transmission was halted MUST be reflected in
the RTP timestamp of each RTP stream. The duration can be calculated
for each RTP stream as the time elapsed from when the last RTP packet
was sent before the PAUSE request was received and when the first RTP
packet was sent after the subsequent PLAY request was received. The
duration includes all latency incurred and processing time required to
complete the request.RFC 3550 states that: "the RTP
timestamp for each unit [packet] would be related to the wallclock
time at which the unit becomes current on the virtual presentation
timeline".In order to satisfy the requirements of , the RTP timestamp space needs to increase
continuously with real time. While this is not optimal for stored
media, it is required for RTP and RTCP to function as intended.
Using a continuous RTP timestamp space allows the same timestamp
model for both stored and live media and allows better opportunity
to integrate both types of media under a single control.As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms, and an initial sequence number and timestamp of
zero.The ensuing RTP data stream is depicted below:The client then sends a PAUSE request:20 seconds elapse and then the client sends a PLAY request. In
addition, the server requires 15 ms to process the request:The ensuing RTP data stream is depicted below:First, NPT 10 through 10.3 is played, then a PAUSE is received by
the server. After 20 seconds, a PLAY is received by the server that
takes 15 ms to process. The duration of time for which the session was
paused is reflected in the RTP timestamp of the RTP packets sent after
this PLAY request.A client can use the RTSP Range header and RTP-Info header to map
NPT time of a presentation with the RTP timestamp.Note: in RFC 2326 , this matter was not
clearly defined and was misunderstood commonly. However, for RTSP 2.0,
it is expected that this will be handled correctly and no exception
handling will be required.Note further: it may be required to reset some of the state to
ensure the correct media decoding and the usual jitter-buffer handling
when issuing a PLAY request.For certain data types, tight integration between the RTSP layer
and the RTP layer will be necessary. This by no means precludes the
above restrictions. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek or before or after a PAUSE.For scaling (see ), RTP timestamps should
correspond to the rendering timing. For example, when playing video
recorded at 30 frames per second at a scale of two and speed () of one, the server would drop every second frame
to maintain and deliver video packets with the normal timestamp
spacing of 3,000 per frame, but NPT would increase by 1/15 second for
each video frame.Note: the above scaling puts requirements on the media codec or
a media stream to support it. For example, motion JPEG or other
non-predictive video coding can easier handle the above
example.The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info ()
header provides the first sequence number of the next segment.For continuous audio, the server SHOULD set the RTP marker bit at
the beginning of serving a new PLAY request or at jumps in timeline.
This allows the client to perform playout delay adaptation.Note that more than one SSRC MAY be sent in the media stream. If it
happens, all sources are expected to be rendered simultaneously.The RTCP BYE message indicates the end of use of a given SSRC. If
all sources leave an RTP session, it can, in most cases, be assumed to
have ended. Therefore, a client or server MUST NOT send an RTCP BYE
message until it has finished using a SSRC. A server SHOULD keep using
an SSRC until the RTP session is terminated. Prolonging the use of a
SSRC allows the established synchronization context associated with
that SSRC to be used to synchronize subsequent PLAY requests even if
the PLAY response is late.An SSRC collision with the SSRC that transmits media does also have
consequences, as it will normally force the media sender to change its
SSRC in accordance with the RTP specification . However, an RTSP server may wait and see if the
client changes and thus resolve the conflict to minimize the impact.
As media sender, SSRC change will result in a loss of synchronization
context and require any receiver to wait for RTCP sender reports for
all media requiring synchronization before being able to play out
synchronized. Due to these reasons, a client joining a session should
take care not to select the same SSRC(s) as the server indicates in
the ssrc Transport header parameter. Any SSRC signaled in the
Transport header MUST be avoided. A client detecting a collision prior
to sending any RTP or RTCP messages SHALL also select a new SSRC.It is the intention that any future protocol or profile regarding
media delivery and lower transport should be easy to add to RTSP. This
section provides the necessary steps that need to be met.The following things need to be considered when adding a new
protocol or profile for use with RTSP: The protocol or profile needs to define a name tag representing
it. This tag is required to be an ABNF "token" to be possible to
use in the Transport header specification.The useful combinations of protocol, profiles, and lower-layer
transport for this extension need to be defined. For each
combination, declare the necessary parameters to use in the
Transport header.For new media protocols, the interaction with RTSP needs to be
addressed. One important factor will be the media synchronization.
It may be necessary to have new headers similar to RTP info to
carry this information.Discussion needs to occur regarding congestion control for
media, especially if transport without built-in congestion control
is used.See the IANA Considerations section () for
information on how to register new attributes.The Session Description Protocol (SDP, ) may
be used to describe streams or presentations in RTSP. This description
is typically returned in reply to a DESCRIBE request on a URI from a
server to a client or received via HTTP from a server to a client.This appendix describes how an SDP file determines the operation of
an RTSP session. Thus, it is worth pointing out that the interpretation
of the SDP is done in the context of the SDP receiver, which is the one
being configured. This is the same as in SAP; this differs from SDP Offer/Answer where each SDP is interpreted
in the context of the agent providing it.SDP as is provides no mechanism by which a client can distinguish,
without human guidance, between several media streams to be rendered
simultaneously and a set of alternatives (e.g., two audio streams spoken
in different languages). The SDP extension found in "The Session
Description Protocol (SDP) Grouping Framework"
provides such functionality to some degree. describes the usage of SDP media line
grouping for RTSP.The terms "session-level", "media-level", and other key/attribute
names and values used in this appendix are to be used as defined in
SDP:The "a=control" attribute is used to convey the control URI. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URI to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URI for aggregate control
(presentation URI). The session-level URI MUST be different from any
media-level URI. The presence of a session-level control attribute
MUST be interpreted as support for aggregated control. The control
attribute MUST be present on the media level unless the presentation
only contains a single media stream; in which case, the attribute
MAY be present on the session level only and then also apply to that
single media stream.ABNF for the attribute is defined in .Example:This attribute MAY contain either relative or absolute URIs,
following the rules and conventions set out in RFC 3986 . Implementations MUST look for a base URI in the
following order: the RTSP Content-Base field;the RTSP Content-Location field;the RTSP Request-URI.If this attribute contains only an asterisk (*), then the
URI MUST be treated as if it were an empty embedded URI; thus, it
will inherit the entire base URI.Note: RFC 2326 was very unclear on the processing of relative
URIs and several RTSP 1.0 implementations at the point of
publishing this document did not perform RFC 3986 processing to
determine the resulting URI; instead, simple concatenation is
common. To avoid this issue completely, it is recommended to use
absolute URIs in the SDP.The URI handling for SDPs from container files needs
special consideration. For example, let's assume that a container
file has the URI: "rtsp://example.com/container.mp4". Let's further
assume this URI is the base URI and that there is an absolute
media-level URI: "rtsp://example.com/container.mp4/trackID=2". A
relative media-level URI that resolves in accordance with RFC 3986
to the above given media URI is
"container.mp4/trackID=2". It is usually not desirable to need to
include or modify the SDP stored within the container file with the
server local name of the container file. To avoid this, one can
modify the base URI used to include a trailing slash, e.g.,
"rtsp://example.com/container.mp4/". In this case, the relative URI
for the media will only need to be "trackID=2". However, this will
also mean that using "*" in the SDP will result in the control URI
including the trailing slash, i.e.,
"rtsp://example.com/container.mp4/".Note: the usage of TrackID in the above is not a standardized
form, but one example out of several similar strings such as
TrackID, Track_ID, StreamID that is used by different server
vendors to indicate a particular piece of media inside a
container file.The "m=" field is used to enumerate the streams. It is expected
that all the specified streams will be rendered with appropriate
synchronization. If the session is over multicast, the port number
indicated SHOULD be used for reception. The client MAY try to
override the destination port, through the Transport header. The
servers MAY allow this: the response will indicate whether or not
this is allowed. If the session is unicast, the port numbers are the
ones RECOMMENDED by the server to the client, about which receiver
ports to use; the client MUST still include its receiver ports in
its SETUP request. The client MAY ignore this recommendation. If the
server has no preference, it SHOULD set the port number value to
zero.The "m=" lines contain information about which transport
protocol, profile, and possibly lower-layer are to be used for the
media stream. The combination of transport, profile, and lower
layer, like RTP/AVP/UDP, needs to be defined for how to be used with
RTSP. The currently defined combinations are discussed in ; further combinations MAY be specified.Example:The payload type or types are specified in the "m=" line. In case
the payload type is a static payload type from RFC 3551 , no other information may be required. In case it
is a dynamic payload type, the media attribute "rtpmap" is used to
specify what the media is. The "encoding name" within the "rtpmap"
attribute may be one of those specified in ,
a media type registered with IANA according to , or an experimental encoding as specified in
SDP). Codec-specific parameters are
not specified in this field, but rather in the "fmtp" attribute
described below.The selection of the RTP payload type numbers used may be
required to consider RTP and RTCP
Multiplexing, if that is to be supported by the server.Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) to which the attribute refers. Note that some of the
format-specific parameters may be specified outside of the "fmtp"
parameters, for example, like the "ptime" attribute for most audio
encodings.The SDP attributes "a=sendrecv", "a=recvonly", and "a=sendonly"
provide instructions about the direction the media streams flow
within a session. When using RTSP, the SDP can be delivered to a
client using either RTSP DESCRIBE or a number of RTSP external
methods, like HTTP, FTP, and email. Based on this, the SDP applies
to how the RTSP client will see the complete session. Thus, media
streams delivered from the RTSP server to the client would be given
the "a=recvonly" attribute."a=recvonly" in an SDP provided to the RTSP client indicates that
media delivery will only occur in the direction from the RTSP server
to the client. SDP provided to the RTSP client that lacks any of the
directionality attributes ("a=recvonly", "a=sendonly", "a=sendrecv")
would be interpreted as having "a=sendrecv". At the time of writing,
there exists no RTSP mode suitable for media traffic in the
direction from the RTSP client to the server. Thus, all RTSP SDP
SHOULD have an "a=recvonly" attribute when using the PLAY mode
defined in this document. If future modes are defined for media in
the client-to-server direction, then usage of "a=sendonly" or
"a=sendrecv" may become suitable to indicate intended media
directions.The "a=range" attribute defines the total time range of the
stored session or an individual media. Live sessions that are not
seekable can be indicated as specified below; whereas the length of
live sessions can be deduced from the "t=" and "r=" SDP
parameters.The attribute is both a session- and a media-level attribute. For
presentations that contain media streams of the same duration, the
range attribute SHOULD only be used at the session level. In case of
different lengths, the range attribute MUST be given at media level
for all media and SHOULD NOT be given at the session level. If the
attribute is present at both media level and session level, the
media-level values MUST be used.Note: usually one will specify the same length for all media,
even if there isn't media available for the full duration on all
media. However, that requires that the server accept PLAY requests
within that range.Servers MUST take care to provide RTSP Range (see ) values that are consistent with what is
presented in the SDP for the content. There is no reason for non
dynamic content, like media clips provided on demand to have
inconsistent values. Inconsistent values between the SDP and the
actual values for the content handled by the server is likely to
generate some failure, like 457 "Invalid Range", in case the client
uses PLAY requests with a Range header. In case the content is
dynamic in length and it is infeasible to provide a correct value in
the SDP, the server is recommended to describe this as content that
is not seekable (see below). The server MAY override that property
in the response to a PLAY request using the correct values in the
Range header.The unit is specified first, followed by the value range. The
units and their values are as defined in ,
, and and MAY be
extended with further formats. Any open-ended range (start-), i.e.,
without stop range, is of unspecified duration and MUST be
considered as content that is not seekable unless this property is
overridden. Multiple instances carrying different clock formats MAY
be included at either session or media level.ABNF for the attribute is defined in .Examples:The "t=" field defines when the SDP is valid. For on-demand
content, the server SHOULD indicate a stop time value for which it
guarantees the description to be valid and a start time that is
equal to or before the time at which the DESCRIBE request was
received. It MAY also indicate start and stop times of 0, meaning
that the session is always available.For sessions that are of live type, i.e., specific start time,
unknown stop time, likely not seekable, the "t=" and "r=" field
SHOULD be used to indicate the start time of the event. The stop
time SHOULD be given so that the live event will have ended at that
time, while still not being unnecessary far into the future.In SDP used with RTSP, the "c=" field contains the destination
address for the media stream. If a multicast address is specified,
the client SHOULD use this address in any SETUP request as
destination address, including any additional parameters, such as
TTL. For on-demand unicast streams and some multicast streams, the
destination address MAY be specified by the client via the SETUP
request, thus overriding any specified address. To identify streams
without a fixed destination address, where the client is required to
specify a destination address, the "c=" field SHOULD be set to a
null value. For addresses of type "IP4", this value MUST be
"0.0.0.0"; and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0"
(can also be written as "::"), i.e., the unspecified address
according to RFC 4291 .The optional "a=mtag" attribute identifies a version of the
session description. It is opaque to the client. SETUP requests may
include this identifier in the If-Match field (see ) to allow session establishment only if this
attribute value still corresponds to that of the current
description. The attribute value is opaque and may contain any
character allowed within SDP attribute values.ABNF for the attribute is defined in .Example:One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session
description types other than SDP for the same piece of media
content.If a presentation does not support aggregate control, no
session-level "a=control" attribute is specified. For an SDP with
multiple media sections specified, each section will have its own
control URI specified via the "a=control" attribute.Example:Note that the position of the control URI in the description
implies that the client establishes separate RTSP control sessions to
the servers audio.example.com and video.example.com.It is recommended that an SDP file contain the complete
media-initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control" attribute, which is used to specify the stream URIs, and a
session-level "a=control" attribute, which is used as the Request-URI
for aggregate control. If the media-level URI is relative, it is
resolved to absolute URIs according to above.Example: In this example, the client is recommended to establish a single
RTSP session to the server, and it uses the URIs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URI rtsp://example.com/movie/, which is
resolved from the "*", controls the whole presentation (movie).A client is not required to issue SETUP requests for all streams
within an aggregate object. Servers should allow the client to ask for
only a subset of the streams.For some types of media, it is desirable to express a relationship
between various media components, for instance, for lip
synchronization or Scalable Video Codec (SVC) . This relationship is expressed on the SDP level by
grouping of media lines, as described in , and
can be exposed to RTSP.For RTSP, it is mainly important to know how to handle grouped
media received by means of SDP, i.e., if the media are under aggregate
control (see ) or if aggregate
control is not available (see ).It is RECOMMENDED that grouped media are handled by aggregate
control, to give the client the ability to control either the whole
presentation or single media.There are some considerations that need to be made when the session
description is delivered to the client outside of RTSP, for example
via HTTP or email.First of all, the SDP needs to contain absolute URIs, since
relative will, in most cases, not work as the delivery will not
correctly forward the base URI.The writing of the SDP session availability information, i.e., "t="
and "r=", needs to be carefully considered. When the SDP is fetched by
the DESCRIBE method, the probability that it is valid is very high.
However, the same is much less certain for SDPs distributed using
other methods. Therefore, the publisher of the SDP should take care to
follow the recommendations about availability in the SDP specification
in Section 4.2.This appendix describes the most important and considered use cases
for RTSP. They are listed in descending order of importance in regard to
ensuring that all necessary functionality is present. This specification
only fully supports usage of the two first. Also, in these first two
cases, there are special cases or exceptions that are not supported
without extensions, e.g., the redirection of media delivery to an
address other than the controlling agent's (client's).An RTSP-capable server stores content suitable for being streamed
to a client. A client desiring playback of any of the stored content
uses RTSP to set up the media transport required to deliver the
desired content. RTSP is then used to initiate, halt, and manipulate
the actual transmission (playout) of the content. RTSP is also
required to provide the necessary description and synchronization
information for the content.The above high-level description can be broken down into a number
of functions of which RTSP needs to be capable. Provide initialization
information about the presentation (content); for example, which
media codecs are needed for the content. Other information that is
important includes the number of media streams the presentation
contains, the transport protocols used for the media streams, and
identifiers for these media streams. This information is required
before setup of the content is possible and to determine if the
client is even capable of using the content. This information need not be sent using RTSP;
other external protocols can be used to transmit the transport
presentation descriptions. Two good examples are the use of HTTP
or email to fetch or receive presentation
descriptions like SDP Set up some or all of the media streams in a
presentation. The setup itself consists of selecting the protocol
for media transport and the necessary parameters for the protocol,
like addresses and ports.After the necessary media
streams have been established, the client can request the server
to start transmitting the content. The client must be allowed to
start or stop the transmission of the content at arbitrary times.
The client must also be able to start the transmission at any
point in the timeline of the presentation.For media-transport protocols like
RTP , it might be beneficial to carry
synchronization information within RTSP. This may be due to either
the lack of inter-media synchronization within the protocol itself
or the potential delay before the synchronization is established
(which is the case for RTP when using RTCP).Terminate the established contexts. For this use case, there are a number of assumptions about
how it works. These are: The content is stored at the
server and can be accessed at any time during a time period when
it is intended to be available.A server is capable of serving
a number of clients simultaneously, including from the same piece
of content at different points in that presentations timeline.Content for each individual
client is transmitted to them using unicast traffic. It is also possible to redirect the media traffic to a
different destination than that of the agent controlling the traffic.
However, allowing this without appropriate mechanisms for checking
that the destination approves of this allows for Distributed DoS
(DDoS).This use case is similar to the above on-demand content case (see
), the difference is the nature
of the content itself. Live content is continuously distributed as it
becomes available from a source; i.e., the main difference from
on-demand is that one starts distributing content before the end of it
has become available to the server.In many cases, the consumer of live content is only interested in
consuming what actually happens "now"; i.e., very similar to broadcast
TV. However, in this case, it is assumed that there exists no
broadcast or multicast channel to the users, and instead the server
functions as a distribution node, sending the same content to multiple
receivers, using unicast traffic between server and client. This
unicast traffic and the transport parameters are individually
negotiated for each receiving client.Another aspect of live content is that it often has a very limited
time of availability, as it is only available for the duration of the
event the content covers. An example of such live content could be a
music concert that lasts two hours and starts at a predetermined time.
Thus, there is a need to announce when and for how long the live
content is available.In some cases, the server providing live content may be saving some
or all of the content to allow clients to pause the stream and resume
it from the paused point, or to "rewind" and play continuously from a
point earlier than the live point. Hence, this use case does not
necessarily exclude playing from other than the live point of the
stream, playing with scales other than 1.0, etc.It is possible to use RTSP to request that media be delivered to a
multicast group. The entity setting up the session (the controller)
will then control when and what media is delivered to the group. This
use case has some potential for DoS attacks by flooding a multicast
group. Therefore, a mechanism is needed to indicate that the group
actually accepts the traffic from the RTSP server.An open issue in this use case is how one ensures that all
receivers listening to the multicast or broadcast receives the session
presentation configuring the receivers. This specification has to rely
on an external solution to solve this issue.If one has an established conference or group session, it is
possible to have an RTSP server distribute media to the whole group.
Transmission to the group is simplest when controlled by a single
participant or leader of the conference. Shared control might be
possible, but would require further investigation and possibly
extensions.This use case assumes that there exists either a multicast or a
conference focus that redistributes media to all participants.This use case is intended to be able to handle the following
scenario: a conference leader or participant (hereafter called the
"controller") has some pre-stored content on an RTSP server that he
wants to share with the group. The controller sets up an RTSP session
at the streaming server for this content and retrieves the session
description for the content. The destination for the media content is
set to the shared multicast group or conference focus. When desired by
the controller, he/she can start and stop the transmission of the
media to the conference group.There are several issues with this use case that are not solved by
this core specification for RTSP: To avoid an RTSP server from being an unknowing
participant in a DoS attack, the server needs to be able to verify
the destination's acceptance of the media. Such a mechanism to
verify the approval of received media does not yet exist; instead,
only policies can be used, which can be made to work in controlled
environments.To
enable a media receiver to correctly decode the content, the media
configuration information needs to be distributed reliably to all
participants. This will most likely require support from an
external protocol.If it is desired to
pass control of the RTSP session between the participants, some
support will be required by an external protocol to exchange state
information and possibly floor control of who is controlling the
RTSP session.This use case in its simplest form does not require any use of RTSP
at all; this is what multicast conferences being announced with SAP and SDP are intended to handle. However,
in use cases where more advanced features like access control to the
multicast session are desired, RTSP could be used for session
establishment.A client desiring to join a live multicasted media session with
cryptographic (encryption) access control could use RTSP in the
following way. The source of the session announces the session and
gives all interested an RTSP URI. The client connects to the server
and requests the presentation description, allowing configuration for
reception of the media. In this step, it is possible for the client to
use secured transport and any desired level of authentication; for
example, for billing or access control. An RTSP link also allows for
load balancing between multiple servers.If these were the only goals, they could be achieved by simply
using HTTP. However, for cases where the sender likes to keep track of
each individual receiver of a session, and possibly use the session as
a side channel for distributing key-updates or other information on a
per-receiver basis, and the full set of receivers is not known prior
to the session start, the state establishment that RTSP provides can
be beneficial. In this case, a client would establish an RTSP session
for this multicast group with the RTSP server. The RTSP server will
not transmit any media, but instead will point to the multicast group.
The client and server will be able to keep the session alive for as
long as the receiver participates in the session thus enabling, for
example, the server to push updates to the client.This use case will most likely not be able to be implemented
without some extensions to the server-to-client push mechanism. Here
the PLAY_NOTIFY method (see ) with a
suitable extension could provide clear benefits.A resource of type "text/parameters" consists of either 1) a list of
parameters (for a query) or 2) a list of parameters and associated
values (for a response or setting of the parameter). Each entry of the
list is a single line of text. Parameters are separated from values by a
colon. The parameter name MUST only use US-ASCII visible characters
while the values are UTF-8 text strings. The media type registration
form is in .There is a potential interoperability issue for this format. It was
named in RFC 2326 but never defined, even if used in examples that hint
at the syntax. This format matches the purpose and its syntax supports
the examples provided. However, it goes further by allowing UTF-8 in the
value part; thus, usage of UTF-8 strings may not be supported. However,
as individual parameters are not defined, the implementing application
needs to have out-of-band agreement or using feature tag anyway to
determine if the endpoint supports the parameters.The ABNF grammar for "text/parameters"
content is:This appendix provides guidance for those who want to implement RTSP
messages over unreliable transports as has been defined in RTSP 1.0. RFC 2326 defined the "rtspu" URI
scheme and provided some basic information for the transport of RTSP
messages over UDP. The information is being provided here as there has
been at least one commercial implementation and compatibility with that
should be maintained.The following points should be considered for an interoperable
implementation:Requests shall be acknowledged by the receiver. If there is no
acknowledgement, the sender may resend the same message after a
timeout of one round-trip time (RTT). Any retransmissions due to
lack of acknowledgement must carry the same sequence number as the
original request.The RTT can be estimated as in TCP (RFC 6298) , with an initial round-trip value of 500 ms. An
implementation may cache the last RTT measurement as the initial
value for future connections.The Timestamp header () is used to
avoid the retransmission ambiguity
problem.The registered default port for RTSP over UDP for the server is
554.RTSP messages can be carried over any lower-layer transport
protocol that is 8-bit clean.RTSP messages are vulnerable to bit errors and should not be
subjected to them.Source authentication, or at least validation that RTSP messages
comes from the same entity becomes extremely important, as session
hijacking may be substantially easier for RTSP message transport
using an unreliable protocol like UDP than for TCP.There are two RTSP headers that are primarily intended for being used
by the unreliable handling of RTSP messages and which will be
maintained: CSeq: See . It should be noted that the
CSeq header is also required to match requests and responses
independent whether a reliable or unreliable transport is used.Timestamp: See This section contains notes on issues about backwards compatibility
with clients or servers being implemented according to RFC 2326 . Note that there exists no requirement to implement
RTSP 1.0; in fact, this document recommends against it as it is
difficult to do in an interoperable way.A server implementing RTSP 2.0 MUST include an RTSP-Version of
"RTSP/2.0" in all responses to requests containing RTSP-Version value of
"RTSP/2.0". If a server receives an RTSP 1.0 request, it MAY respond
with an RTSP 1.0 response if it chooses to support RFC 2326. If the
server chooses not to support RFC 2326, it MUST respond with a 505 (RTSP
Version Not Supported) status code. A server MUST NOT respond to an RTSP
1.0 request with an RTSP 2.0 response.Clients implementing RTSP 2.0 MAY use an OPTIONS request with an
RTSP-Version of "RTSP/2.0" to determine whether a server supports RTSP
2.0. If the server responds with either an RTSP-Version of "RTSP/1.0" or
a status code of 505 (RTSP Version Not Supported), the client will have
to use RTSP 1.0 requests if it chooses to support RFC 2326.The behavior in the server when a Play is received in Play state
has changed (). In RFC 2326, the new PLAY
request would be queued until the current Play completed. Any new PLAY
request now takes effect immediately replacing the previous
request.Some server implementations of RFC 2326 maintain a one-to-one
relationship between a connection and an RTSP session. Such
implementations require clients to use a persistent connection to
communicate with the server and when a client closes its connection,
the server may remove the RTSP session. This is worth noting if an
RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.This appendix briefly lists the differences between RTSP 1.0 and RTSP 2.0 for an informational
purpose. For implementers of RTSP 2.0, it is recommended to read
carefully through this memo and not to rely on the list of changes below
to adapt from RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be
backwards compatible with RTSP 1.0 other
than the version negotiation mechanism.The following protocol elements were removed in RTSP 2.0 compared
to RTSP 1.0:the RECORD and ANNOUNCE methods and all related functionality
(including 201 (Created) and 250 (Low On Storage Space) status
codes);the use of UDP for RTSP message transport (due to missing
interest and to broken specification);the use of PLAY method for keep-alive in Play state.The following protocol elements were added or changed in RTSP 2.0
compared to RTSP 1.0:RTSP session TEARDOWN from the server to the client;IPv6 support;extended IANA registries (e.g., transport headers parameters,
transport-protocol, profile, lower-transport, and mode);request pipelining for quick session start-up;fully reworked state machine;RTSP messages now use URIs rather than URLs;incorporated much of related HTTP text () in this memo, compared to just referencing the
sections in HTTP, to avoid ambiguities;the REDIRECT method was expanded and diversified for different
situations;Includes a new section about how to set up different
media-transport alternatives and their profiles in addition to
lower-layer protocols. This caused the appendix on RTP interaction
to be moved to the new section instead of being in the part that
describes RTP. The section also includes guidelines what to
consider when writing usage guidelines for new protocols and
profiles;Added an asynchronous notification method PLAY_NOTIFY. This
method is used by the RTSP server to asynchronously notify clients
about session changes while in Play state. To a limited extent,
this is comparable with some implementations of ANNOUNCE in RTSP
1.0 not intended for Recording.The below changes have been made to RTSP 1.0 (RFC 2326) when
defining RTSP 2.0. Note that this list does not reflect minor changes
in wording or correction of typographical errors. The section on minimal implementation was deleted. Instead, the
main part of the specification defines the core of RTSP 2.0.The Transport header has been changed in the following ways:
The ABNF has been changed to define that extensions are
possible and that unknown parameters result in servers
ignoring the transport specification.To prevent backwards compatibility issues, any extension or
new parameter requires the usage of a feature tag combined
with the Require header.Syntax ambiguities with the Mode parameter have been
resolved.Syntax error with ";" for multicast and unicast has been
resolved.Two new addressing parameters have been defined: src_addr
and dest_addr. These replace the parameters "port",
"client_port", "server_port", "destination", and "source".Support for IPv6 explicit addresses in all address fields
has been included.To handle URI definitions that contain ";" or ",", a
quoted-URI format has been introduced and is required.IANA registries for the transport header parameters,
transport-protocol, profile, lower-transport, and mode have
been defined.The Transport header's interleaved parameter's text was
made more strict and uses formal requirements levels. It was
also clarified that the interleaved channels are symmetric and
that it is the server that sets the channel numbers.It has been clarified that the client can't request of the
server to use a certain RTP SSRC, using a request with the
transport parameter SSRC.Syntax definition for SSRC has been clarified to require
8HEX. It has also been extended to allow multiple values for
clients supporting this version.Clarified the text on the Transport header's "dest_addr"
parameters regarding what security precautions the server is
required to perform.The Range formats have been changed in the following way: The NPT format has been given an initial NPT identifier
that must now be used.All formats now support initial open-ended formats of type
"npt=-10" and also format only "Range: smpte" ranges for usage
with GET_PARAMETER requests.The npt-hhmmss notation now follows ISO 8601 more
strictly.RTSP message handling has been changed in the following ways:
RTSP messages now use URIs rather than URLs.It has been clarified that a 4xx message due to a missing
CSeq header shall be returned without a CSeq header.The 300 (Multiple Choices) response code has been
removed.Rules for how to handle the timing out RTSP messages have
been added.Extended Pipelining rules allowing for quick session
startup.Sequence numbering and proxy handling of sequence numbers
have been defined, including cases when responses arrive out
of order.The HTTP references have been updated to first RFCs 2616 and
2617 and then to RFC 7230-7235. Most of the text has been copied
and then altered to fit RTSP into this specification. The Public
and the Content-Base headers have also been imported from RFC 2068
so that they are defined in the RTSP specification. Known effects
on RTSP due to HTTP clarifications: Content-Encoding header can include encoding of type
"identity".The state machine section has been completely rewritten. It now
includes more details and is also more clear about the model
used.An IANA section has been included that contains a number of
registries and their rules. This will allow us to use IANA to keep
track of RTSP extensions.The transport of RTSP messages has seen the following changes:
The use of UDP for RTSP message transport has been
deprecated due to missing interest and to broken
specification.The rules for how TCP connections are to be handled have
been clarified. Now it is made clear that servers should not
close the TCP connection unless they have been unused for
significant time.Strong recommendations why servers and clients should use
persistent connections have also been added.There is now a requirement on the servers to handle
non-persistent connections as this provides fault
tolerance.Added wording on the usage of Connection:Close for
RTSP.Specified usage of TLS for RTSP messages, including a
scheme to approve a proxy's TLS connection to the next
hop.The following header-related changes have been made: Accept-Ranges response-header has been added. This header
clarifies which range formats can be used for a resource.Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-seconds
for max-stale and min-fresh parameters.Put requirement on CSeq header that the value is increased
by one for each new RTSP request. A recommendation to start at
0 has also been added.Added a requirement that the Date header must be used for
all messages with a message body and the Server should always
include it.Removed the possibility of using Range header with Scale
header to indicate when it is to be activated, since it can't
work as defined. Also, added a rule that lack of Scale header
in a response indicates lack of support for the header.
feature tags for scaled playback have been defined.The Speed header must now be responded to in order to
indicate support and the actual speed going to be used. A
feature tag is defined. Notes on congestion control were also
added.The Supported header was borrowed from SIP to help with the feature
negotiation in RTSP.Clarified that the Timestamp header can be used to resolve
retransmission ambiguities.The Session header text has been expanded with an
explanation on keep-alive and which methods to use.
SET_PARAMETER is now recommended to use if only keep-alive
within RTSP is desired.It has been clarified how the Range header formats are used
to indicate pause points in the PAUSE response.Clarified that RTP-Info URIs that are relative use the
Request-URI as base URI. Also clarified that the used URI must
be the one that was used in the SETUP request. The URIs are
now also required to be quoted. The header also expresses the
SSRC for the provided RTP timestamp and sequence number
values.Added text that requires the Range to always be present in
PLAY responses. Clarified what should be sent in case of live
streams.The headers table has been updated using a structure
borrowed from SIP. Those tables convey much more information
and should provide a good overview of the available
headers.It has been clarified that any message with a message body
is required to have a Content-Length header. This was the case
in RFC 2326, but could be misinterpreted.ETag has changed its name to MTag.To resolve functionality around MTag, the MTag and
If-None-Match header have been added from HTTP with necessary
clarification in regard to RTSP operation.Imported the Public header from HTTP (RFC 2068 ) since it has been removed from HTTP due to
lack of use. Public is used quite frequently in RTSP.Clarified rules for populating the Public header so that it
is an intersection of the capabilities of all the RTSP agents
in a chain.Added the Media-Range header for listing the current
availability of the media range.Added the Notify-Reason header for giving the reason when
sending PLAY_NOTIFY requests.A new header Seek-Style has been defined to direct and
inform how any seek operation should/have been performed.The Protocol Syntax has been changed in the following way:
All ABNF definitions are updated according to the rules
defined in RFC 5234 and have been
gathered in a separate section ().The ABNF for the User-Agent and Server headers have been
corrected.Some definitions in the introduction regarding the RTSP
session have been changed.The protocol has been made fully IPv6 capable.The CHAR rule has been changed to exclude NULL.The Status codes have been changed in the following ways: The use of status code 303 (See Other) has been deprecated
as it does not make sense to use in RTSP.The never-defined status code 411 "Length Required" has
been completely removed.When sending response 451 (Parameter Not Understood) and
458 (Parameter Is Read-Only), the response body should contain
the offending parameters.Clarification on when a 3rr redirect status code can be
received has been added. This includes receiving 3rr as a
result of a request within an established session. This
provides clarification to a previous unspecified behavior.Removed the 201 (Created) and 250 (Low On Storage Space)
status codes as they are only relevant to recording, which is
deprecated.Several new status codes have been defined: 464 (Data
Transport Not Ready Yet), 465 (Notification Reason Unknown),
470 (Connection Authorization Required), 471 (Connection
Credentials Not Accepted), and 472 (Failure to Establish
Secure Connection).The following functionality has been deprecated from the
protocol: The use of Queued Play.The use of PLAY method for keep-alive in Play state.The RECORD and ANNOUNCE methods and all related
functionality. Some of the syntax has been removed.The possibility to use timed execution of methods with the
time parameter in the Range header.The description on how rtspu works is not part of the core
specification and will require external description. Only that
it exists is mentioned here and some requirements for the
transport are provided.The following changes have been made in relation to methods:
The OPTIONS method has been clarified with regard to the
use of the Public and Allow headers.Added text clarifying the usage of SET_PARAMETER for
keep-alive and usage without a body.PLAY method is now allowed to be pipelined with the
pipelining of one or more SETUP requests following the initial
that generates the session for aggregated control.REDIRECT has been expanded and diversified for different
situations.Added a new method PLAY_NOTIFY. This method is used by the
RTSP server to asynchronously notify clients about session
changes.Wrote a new section about how to set up different
media-transport alternatives and their profiles as well as
lower-layer protocols. This caused the appendix on RTP interaction
to be moved to the new section instead of being in the part that
describes RTP. The new section also includes guidelines what to
consider when writing usage guidelines for new protocols and
profiles.Setup and usage of independent TCP connections for transport of
RTP has been specified.Added a new section describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature tags.Added a Contributors section with people who have contributed
actual text to the specification.Added a section "Use Cases" that describes the major use cases
for RTSP.Clarified the usage of a=range and how to indicate live content
that are not seekable with this header.Text specifying the special behavior of PLAY for live
content.Security features of RTSP have been clarified:HTTP-based authorization has been clarified requiring both
Basic and Digest supportTLS support has been mandatedIf one implements RTP, then SRTP and defined MIKEY-based
key-exchange must be supportedVarious minor mitigations discussed or resulted in protocol
changes.This memorandum defines RTSP version 2.0, which is a revision of the
Proposed Standard RTSP version 1.0 defined in .
The authors of RFC 2326 are Henning Schulzrinne, Anup Rao, and Robert
Lanphier.Both RTSP version 1.0 and RTSP version 2.0 borrow format and
descriptions from HTTP/1.1.Robert Sparks and especially Elwyn Davies provided very valuable and
detailed reviews in the IETF Last Call that greatly improved the
document and resolved many issues, especially regarding consistency.This document has benefited greatly from the comments of all those
participating in the MMUSIC WG. In addition to those already mentioned,
the following individuals have contributed to this specification:Rahul Agarwal, Claudio Allocchio, Jeff Ayars, Milko Boic, Torsten
Braun, Brent Browning, Bruce Butterfield, Steve Casner, Maureen Chesire,
Jinhang Choi, Francisco Cortes, Elwyn Davies, Spencer Dawkins, Kelly
Djahandari, Martin Dunsmuir, Adrian Farrel, Stephen Farrell, Ross
Finlayson, Eric Fleischman, Jay Geagan, Andy Grignon, Christian Groves,
V. Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
John K. Ho, Patrick Hoffman, Go Hori, Philipp Hoschka, Anne Jones,
Ingemar Johansson, Jae-Hwan Kim, Anders Klemets, Ruth Lang, Barry Leiba,
Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Chris Lonvick, Xavier
Marjou, Thomas Marshall, Rob McCool, Martti Mela, David Oran, Joerg Ott,
Joe Pallas, Maria Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan,
Colin Perkins, Pekka Pessi, Igor Plotnikov, Pete Resnick, Peter
Saint-Andre, Holger Schmidt, Jonathan Sergent, Pinaki Shah, David
Singer, Lior Sion, Jeff Smith, Alexander Sokolsky, Dale Stammen, John
Francis Stracke, Geetha Srikantan, Scott Taylor, David Walker, Stephan
Wenger, Dale R. Worley, and Byungjo Yoon, and especially Flemming
Andreasen.The following people have made written contributions that were
included in the specification: Tom Marshall contributed text on the usage of 3rr status
codes.Thomas Zheng contributed text on the usage of the Range in PLAY
responses and proposed an earlier version of the PLAY_NOTIFY
method.Sean Sheedy contributed text on the timeout behavior of RTSP
messages and connections, the 463 (Destination Prohibited) status
code, and proposed an earlier version of the PLAY_NOTIFY method.Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
method.Fredrik Lindholm contributed text about the RTSP security
framework.John Lazzaro contributed the text for RTP over Independent
TCP.Aravind Narasimhan contributed by rewriting "Media-Transport Alternatives" and
making editorial improvements on a number of places in the
specification.Torbjorn Einarsson has done some editorial improvements of the
text.