rfc9071.original.xml   rfc9071.xml 
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<front> <front>
<title abbrev="RTP-mixer format for multiparty RTT">RTP-mixer formatting of <title abbrev="RTP-Mixer Format for Multiparty RTT">RTP-Mixer Formatting of
multiparty Real-time text</title> Multiparty Real-Time Text</title>
<seriesInfo name="Internet-Draft" value="draft-ietf-avtcore-multi-party-rtt- <seriesInfo name="RFC" value="9071"/>
mix-20"/> <author fullname="Gunnar Hellström" initials="G" surname="Hellström">
<author fullname="Gunnar Hellstrom" initials="G" surname="Hellstrom"> <organization abbrev="GHAccess">Gunnar Hellström Accessible Communication<
<organization>Gunnar Hellstrom Accessible Communication</organization> /organization>
<address> <address>
<postal> <postal>
<street></street> <city>Vendelsö</city>
<city>Vendelso</city> <code>13670</code>
<!-- <region/> -->
<code>13670</code>
<country>Sweden</country> <country>Sweden</country>
</postal> </postal>
<phone></phone>
<!-- <facsimile/> -->
<email>gunnar.hellstrom@ghaccess.se</email> <email>gunnar.hellstrom@ghaccess.se</email>
<!-- <uri/> -->
</address> </address>
</author> </author>
<date year="2021"/> <date year="2021" month="June" />
<area>ART</area>
<workgroup>AVTCore</workgroup> <keyword>conference</keyword>
<!-- <keyword/> --> <keyword>bridge</keyword>
<!-- <keyword/> --> <keyword>SIP</keyword>
<!-- <keyword/> -->
<!-- <keyword/> -->
<abstract> <abstract>
<t> <t>
This This document provides enhancements of real-time text (as specified in RFC 4103)
document provides enhancements for RFC 4103 real-time text mixing suitable for a suitable for mixing in a centralized conference model, enabling source identifi
centralized conference model that enables source identification and rapidly cation and rapidly
interleaved transmission of text from different sources. The intended use is for interleaved transmission of text from different sources. The intended use is for
real-time text mixers and participant endpoints capable of providing an efficie real-time text mixers and participant endpoints capable of providing an efficie
nt presentation or other treatment of a multiparty real-time text session. The s nt presentation or other treatment of a multiparty real-time text session. The s
pecified mechanism builds on the standard use of the Contributing Source (CSRC) pecified mechanism builds on the standard use of the Contributing Source (CSRC)
list in the Realtime Protocol (RTP) packet for source identification. The method list in the Real-time Transport Protocol (RTP) packet for source identification.
makes use of the same "text/t140" and "text/red" formats as for two-party sessi The method makes use of the same "text/t140" and "text/red" formats as for two-
ons. party sessions.
</t> </t>
<t> <t>
Solutions using multiple RTP streams in the same RTP session are briefly mention ed, as they could have some benefits over the RTP-mixer model. The possibility t o implement the solution in a wide range of existing RTP implementations made th e RTP-mixer model be selected to be fully specified in this document. Solutions using multiple RTP streams in the same RTP session are briefly mention ed, as they could have some benefits over the RTP-mixer model. The RTP-mixer mod el was selected to be used for the fully specified solution in this document bec ause it can be applied to a wide range of existing RTP implementations.
</t> </t>
<t> <t>
A capability exchange is specified so that it can be verified that a mixer and a participant can handle the multiparty-coded real-time text stream using the RT P-mixer method. The capability is indicated by use of an RFC 8866 Session Descri ption Protocol (SDP) media attribute "rtt-mixer". A capability exchange is specified so that it can be verified that a mixer and a participant can handle the multiparty-coded real-time text stream using the RT P-mixer method. The capability is indicated by the use of a Session Description Protocol (SDP) (RFC 8866) media attribute, "rtt-mixer".
</t> </t>
<t> <t>
The document updates RFC 4103 "RTP Payload for Text Conversation". This document updates RFC 4103 ("RTP Payload for Text Conversation").
</t> </t>
<t> <t>
A specification of how a mixer can format text for the case when the endpoint i s not multiparty-aware is also provided. A specification for how a mixer can format text for the case when the endpoint is not multiparty aware is also provided.
</t> </t>
</abstract> </abstract>
</front> </front>
<middle> <middle>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Introduction</name> <name>Introduction</name>
<t> <t>
"RTP Payload for Text Conversation" <xref target="RFC4103" format="default"/> sp "<xref target="RFC4103" format="title"/>" <xref target="RFC4103" format="default
ecifies use of the Real-Time Transport Protocol (RTP) <xref target="RFC3550" for "/>
mat="default"/> for transmission of real-time text (RTT) and the "text/t140" for specifies the use of the Real-time Transport Protocol (RTP) <xref target="RFC355
mat. It also specifies a redundancy format "text/red" for increased robustness. 0" format="default"/> for transmission of real-time text (often called RTT) and
The "text/red" format is registered in <xref target="RFC4102" format="default"/> the "text/t140" format. It also specifies a redundancy format, "text/red", for i
. ncreased robustness. The "text/red" format is registered in <xref target="RFC410
2" format="default"/>.
</t> </t>
<t> <t>
Real-time text is usually provided together with audio and sometimes with vide o in conversational sessions. Real-time text is usually provided together with audio and sometimes with vide o in conversational sessions.
</t> </t>
<t> <t>
A requirement related to multiparty sessions from the presentation level standar A requirement related to multiparty sessions from the presentation-level standar
d T.140 <xref target="T140" format="default"/> for real-time text is: d T.140 <xref target="T140" format="default"/> for real-time text is as follows:
"The display of text from the members of the conversation should be arranged so </t>
that the text from <!-- DNE; verified -->
<blockquote>The display of text from the members of the conversation should be a
rranged so that the text from
each participant is clearly readable, and its source and the relative timing of entered text is visualized each participant is clearly readable, and its source and the relative timing of entered text is visualized
in the display." in the display.</blockquote>
</t>
<t> <t>
Another requirement is that the mixing procedure must not introduce delays in th e text streams that are experienced to be disturbing the real-time experience of the receiving users. Another requirement is that the mixing procedure must not introduce delays in th e text streams that could be perceived as disruptive to the real-time experience of the receiving users.
</t> </t>
<t> <t>
Use of RTT is increasing, and specifically, use in emergency calls is increasing . Emergency call use requires multiparty mixing because it is common that one a gent needs to transfer the call to another specialized agent but is obliged to s tay on the call at least to verify that the transfer was successful. Mixer imple mentations for RFC 4103 "RTP Payload for Text Conversation" can use traditional RFC 3550 RTP functions for mixing and source identification, but the performance of the mixer when giving turns for the different sources to transmit is limited when using the default transmission characteristics with redundancy. The use of real-time text is increasing, and specifically, use in emergency call s is increasing. Emergency call use requires multiparty mixing, because it is c ommon that one agent needs to transfer the call to another specialized agent but is obliged to stay on the call to at least verify that the transfer was success ful. Mixer implementations for RFC 4103 ("RTP Payload for Text Conversation") ca n use traditional RTP functions (RFC 3550) for mixing and source identification, but the performance of the mixer when giving turns for the different sources to transmit is limited when using the default transmission characteristics with re dundancy.
</t> </t>
<t> <t>
The redundancy scheme of <xref target="RFC4103" format="default"/> enables effic ient transmission of earlier transmitted redundant text in packets together with new text. However, the redundancy header format has no source indicators for th e redundant transmissions. The redundant parts in a packet must therefore be fro m the same source as the new text. The recommended transmission is one new and t wo redundant generations of text (T140blocks) in each packet and the recommended transmission interval for two-party use is 300 ms. The redundancy scheme described in <xref target="RFC4103" format="default"/> ena bles efficient transmission of earlier transmitted redundant text in packets tog ether with new text. However, the redundancy header format has no source indicat ors for the redundant transmissions. The redundant parts in a packet must theref ore be from the same source as the new text. The recommended transmission is one new and two redundant generations of text (T140blocks) in each packet, and the recommended transmission interval for two-party use is 300 ms.
</t> </t>
<t> <t>
Real-time text mixers for multiparty sessions need to include the Real-time text mixers for multiparty sessions need to include the
source with each transmitted group of text from a conference participant so that the text can be transmitted interleaved with text groups from different sources at the rate they are created. This enables the text groups to be presented by e ndpoints in suitable grouping with other text from the same source. source with each transmitted group of text from a conference participant so that the text can be transmitted interleaved with text groups from different sources at the rate at which they are created. This enables the text groups to be prese nted by endpoints in a suitable grouping with other text from the same source.
</t> </t>
<t> <t>
The presentation can then be arranged so that text from different sources can be The presentation can then be arranged so that text from different sources can be
presented in real-time and easily read. At the same time it is possible for a r presented in real time and easily read. At the same time, it is possible for a
eading user to perceive approximately when the text was created in real time by reading user to perceive approximately when the text was created in real time by
the different parties. The transmission and mixing is intended to be done in a g the different parties. The transmission and mixing are intended to be done in a
eneral way, so that presentation can be arranged in a layout decided by the endp general way, so that presentation can be arranged in a layout decided upon by t
oint. he receiving endpoint.
</t> </t>
<t> <t>
There are existing implementations of RFC 4103 in endpoints without the updates from this document. These will not be able to receive and present real-time tex t mixed for multiparty-aware endpoints. Existing implementations of RFC 4103 in endpoints that do not implement the upda tes specified in this document cannot be expected to properly present real-time text mixed for multiparty-aware endpoints.
</t> </t>
<t> <t>
A negotiation mechanism is therefore needed for verification if the parti A negotiation mechanism is therefore needed to verify if the parties
es are able to handle a common method for multiparty transmission and agreeing o (1)&nbsp;are able to handle a common method for multiparty transmissions and
n using that method. (2)&nbsp;can agree on using that method.
</t> </t>
<t> <t>
A fallback mixing procedure is also needed for cases when the negotiation result indicates that a receiving endpoint is not capable of handling the mixed format. Multiparty-unaware endpoints would possibly otherwise present all recei ved multiparty mixed text as if it came from the same source regardless of any a ccompanying source indication coded in fields in the packet. Or they may have ot her undesirable ways of acting on the multiparty content. The fallback method is called the mixing procedure for multiparty-unaware endpoints. The fallback meth od is naturally not expected to meet all performance requirements placed on the mixing procedure for multiparty-aware endpoints. A fallback mixing procedure is also needed for cases when the negotiatio n result indicates that a receiving endpoint is not capable of handling the mixe d format. Multiparty-unaware endpoints would possibly otherwise present all rece ived multiparty mixed text as if it came from the same source regardless of any accompanying source indication coded in fields in the packet. Or, they may have other undesirable ways of acting on the multiparty content. The fallback method is called the mixing procedure for multiparty-unaware endpoints. The fallback me thod is naturally not expected to meet all performance requirements placed on th e mixing procedure for multiparty-aware endpoints.
</t> </t>
<t> <t>
The document updates <xref target="RFC4103" format="default"/> by int This document updates <xref target="RFC4103" format="default"/> by i
roducing an attribute for declaring support of the RTP-mixer-based multiparty mi ntroducing an attribute for declaring support of the RTP-mixer-based
xing case and rules for source indications and interleaving of text from differe multiparty-mixing case and rules for source indications and interleaving of text
nt sources. from different sources.
</t> </t>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Terminology</name> <name>Terminology</name>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL <t>The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>",
"MAY", and "OPTIONAL" in this document are to be interpreted as "<bcp14>SHALL NOT</bcp14>", "<bcp14>SHOULD</bcp14>",
described in BCP 14 <xref target="RFC2119"/> <xref target="RFC8174"/> when, and "<bcp14>SHOULD NOT</bcp14>",
only when, they "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
appear in all capitals, as shown above. "<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document
</t> are to be interpreted as described in BCP&nbsp;14
<xref target="RFC2119"/> <xref target="RFC8174"/> when, and only
when, they appear in all capitals, as shown here.</t>
<t> <t>
The terms Source Description (SDES), Canonical name (CNAME), Name (NAME) , Synchronization Source (SSRC), Contributing Source (CSRC), CSRC list, CSRC cou nt [CC], Real-Time control protocol (RTCP), RTP-mixer, RTP-translator are define d in <xref target="RFC3550" format="default"/>. The terms "Source Description" (SDES), "Canonical Name" (CNAME), "Name" (NAME), "Synchronization Source" (SSRC), "Contributing Source" (CSRC), "CSRC li st", "CSRC count" (CC), "RTP Control Protocol" (RTCP), and "RTP mixer" are defin ed in <xref target="RFC3550" format="default"/>.
</t> </t>
<t> "real-time text" (RTT) is text transmitted instantly as it is typed
or created. Recipients can immediately read the message while it is being writt
en, without waiting.
</t>
<t> <t>
The term "T140block" is defined in <xref target="RFC4103" format="defaul t"/> to contain one or more T.140 code elements. The term "T140block" is defined in <xref target="RFC4103" format="defau lt"/> to contain one or more T.140 code elements.
</t> </t>
<t> <t>
"TTY" stands for a textphone type used in North America. "TTY" stands for a textphone type used in North America.
</t> </t>
<t> <t>
Web based real-time communication (WebRTC) is specified by the World Wide Web Consortium (W3C) and IETF. See <xref target="RFC8825" format="default"/>. Web Real-Time Communication (WebRTC) is specified by the World Wide Web Co nsortium (W3C) and the IETF. See <xref target="RFC8825" format="default"/>.
</t> </t>
<t> <t>
"DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension for use with Secure Real-Time Transport Protocol/Secure Real-Time Control Protocol ( SRTP/SRTCP) specified in <xref target="RFC5764" format="default"/>. "DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension for use with the Secure Real-time Transport Protocol / Secure Real-time Transport Co ntrol Protocol (SRTP/SRTCP) as specified in <xref target="RFC5764" format="defau lt"/>.
</t> </t>
<t> <t>
"multiparty-aware" describes an endpoint receiving real-time text from The term "multiparty aware" describes an endpoint that (1) receives
multiple sources through a common conference mixer being able to present the tex real-time text from multiple sources through a common conference
t in real-time, separated by source, and presented so that a user can get an imp mixer, (2) is able to present the text in real time, separated by
ression of the approximate relative timing of text from different parties. source, and (3) presents the text so that a user can get an
</t> impression of the approximate relative timing of text from
<t> different parties.
"multiparty-unaware" describes an endpoint not itself being able to sep </t>
arate text from different sources when received through a common conference mixe <t>
r. The term "multiparty unaware" describes an endpoint that cannot itself
</t> separate text from different sources when the text is received through a common
conference mixer.
</t>
</section> </section>
<section anchor="alternatives" numbered="true" toc="default"> <section anchor="alternatives" numbered="true" toc="default">
<name>Selected solution and considered alternatives</name> <name>Main Method, Fallback Method, and Considered
<t> Alternatives</name>
A number of alternatives were considered when searching an efficient and eas <t>
ily implemented multiparty method for real-time text. This section explains a fe A number of alternatives were considered when searching for an efficient and
w of them briefly. easily implemented multiparty method for real-time text. This section briefly e
xplains a few of them.
</t> </t>
<dl newline="true"> <dl newline="true">
<dt> <dt>
Multiple RTP streams, one per participant Multiple RTP streams, one per participant:
</dt> </dt>
<dd> <dd>
One RTP stream per source would be sent in the same RTP session with the "text/r ed" format. One RTP stream per source would be sent in the same RTP session with the "text/r ed" format.
From some points of view, use of multiple RTP streams, one for each source, s From some points of view, the use of multiple RTP streams, one for each sourc
ent in the same RTP session would be efficient, and would use exactly the same p e, sent in the same RTP session would be efficient and would use exactly the sam
acket format as <xref target="RFC4103" format="default"/> and the same payload t e packet format as <xref target="RFC4103" format="default"/> and the same payloa
ype. d type.
A couple of relevant scenarios using multiple RTP-streams are specified in "R A couple of relevant scenarios using multiple RTP streams are specified in
TP Topologies" <xref target="RFC7667" format="default"/>. One possibility of spe "<xref target="RFC7667" format="title"/>" <xref target="RFC7667" format="default
cial interest is the Selective Forwarding Middlebox (SFM) topology specified in "/>. One possibility of special interest is the Selective Forwarding Middlebox (
RFC 7667 section 3.7 that could enable end-to-end encryption. In contrast to aud SFM) topology specified in
io and video, real-time text is only transmitted when the users actually transmi <xref target="RFC7667" sectionFormat="of" section="3.7"/>,
t information. Thus, an SFM solution would not need to exclude any party from tr which could enable end-to-end encryption. In contrast to audio and video, real-t
ansmission under normal conditions. In order to allow the mixer to convey the pa ime text is only transmitted when the users actually transmit information. Thus,
ckets with the payload preserved and encrypted, an SFM solution would need to ac an SFM solution would not need to exclude any party from transmission under nor
t on some specific characteristics of the "text/red" format. The redundancy head mal conditions. In order to allow the mixer to convey the packets with the paylo
ers are part of the payload, so the receiver would need to just assume that the ad preserved and encrypted, an SFM solution would need to act on some specific c
payload type number in the redundancy header is for "text/t140". The characters haracteristics of the "text/red" format. The redundancy headers are part of the
per second parameter (cps) would need to act per stream. The relation between th payload, so the receiver would need to just assume that the payload type number
e SSRC and the source would need to be conveyed in some specified way, e.g., in in the redundancy header is for "text/t140". The characters per second ("cps") p
the CSRC. Recovery and loss detection would preferably be based on sequence numb arameter would need to act per stream. The relationship between the SSRC and the
er gap detection. Thus, sequence number gaps in the incoming stream to the mixer source would need to be conveyed in some specified way, e.g., in the CSRC. Reco
would need to be reflected in the stream to the participant, with no new gaps c very and loss detection would preferably be based on RTP sequence number gap det
reated by the mixer. ection. Thus, sequence number gaps in the incoming stream to the mixer would nee
However, the RTP implementation in both mixers and endpoints need to support d to be reflected in the stream to the participant, with no new gaps created by
multiple streams in the same RTP session in order to use this mechanism. For bes the mixer.
t deployment opportunity, it should be possible to upgrade existing endpoint sol However, the RTP implementation in both mixers and endpoints needs to support
utions to be multiparty-aware with a reasonable effort. There is currently a lac multiple streams in the same RTP session in order to use this mechanism. To pro
k of support for multi-stream RTP in certain implementations. This fact led to t vide the best opportunities for deployment, it should be possible to upgrade exi
his solution being only briefly mentioned in this document as an option for furt sting endpoint solutions to be multiparty aware with a reasonable amount of effo
her study. rt. There is currently a lack of support for multi-stream RTP in certain impleme
ntations. This fact led to only brief mention of this solution in this document
as an option for further study.
</dd> </dd>
<dt> <dt>
RTP-mixer-based method for multiparty-aware endpoints RTP-mixer-based method for multiparty-aware endpoints:
</dt> </dt>
<dd> <dd>
The "text/red" format in RFC 4103 is sent with a shorter transmission interval The "text/red" format as defined in RFC 4102 and applied in RFC 4103 is sent wi
with the RTP-mixer method and indicating the source in the CSRC field. th the RTP-mixer method indicating the source in the CSRC field.
The "text/red" format with a "text/t140" payload in a single RTP stream can b The "text/red" format with a "text/t140" payload in a single RTP stream can be
e sent when text is available from the call participants instead of at the regul sent when text is available from the call participants instead of at the regular
ar 300 ms. Transmission of packets with text from different sources can then be 300 ms intervals. Transmission of packets with text from different sources can
done smoothly while simultaneous transmission occurs as long as it is not limite then be done smoothly while simultaneous transmission occurs as long as it is no
d by the maximum character rate "cps". With ten participants sending text simult t limited by the maximum character rate "cps" value. With ten participants sendi
aneously, the switching and transmission performance is good. With more simultan ng text simultaneously, the switching and transmission performance is good. With
eously sending participants, and with receivers having the default capacity ther more simultaneously sending participants and with receivers at default capacity
e will be a noticeable jerkiness and delay in text presentation. The jerkiness w , there will be a noticeable jerkiness and delay in text presentation. The more
ill be more expressed the more participants who send text simultaneously. Two se participants who send text
conds jerkiness will be noticeable and slightly unpleasant, but it corresponds i simultaneously, the more jerkiness will occur. Two seconds of jerkiness will be
n time to what typing humans often cause by hesitation or changing position whil noticeable and slightly unpleasant, but it corresponds in time to what typing hu
e typing. A benefit of this method is that no new packet format needs to be intr mans often cause by hesitating or changing position while typing. A benefit of t
oduced and implemented. Since simultaneous typing by more than two parties is ex his method is that no new packet format needs to be introduced and implemented.
pected to be very rare as described in <xref target="intendedapp" format="defaul Since simultaneous typing by more than two parties is expected to be very rare -
t"/>, this method can be used successfully with good performance. Recovery of te - as described in <xref target="intendedapp" format="default"/> -- this method c
xt in case of packet loss is based on analysis of timestamps of received redunda an be used successfully with good performance. Recovery of text in the case of p
ncy versus earlier received text. Negotiation is based on a new SDP media attrib acket loss is based on analysis of timestamps of received redundancy versus earl
ute "rtt-mixer". This method is selected to be the main one specified in this do ier received text. Negotiation is based on a new SDP media attribute, "rtt-mixer
cument. ". This method was selected to be the main method specified in this document.
</dd> </dd>
<dt>Multiple sources per packet</dt> <dt>Multiple sources per packet:</dt>
<dd> <dd>
A new "text" media subtype would be specified with up to 15 sources in ea A new "text" media subtype would be specified with up to 15 sources in e
ch packet. ach packet.
The mechanism would make use of the RTP mixer model specified in RTP <xref ta The mechanism would make use of the RTP-mixer model specified in RTP <xref ta
rget="RFC3550" format="default"/>. The sources are indicated in strict order in rget="RFC3550" format="default"/>. The sources would be indicated in strict orde
the CSRC list of the RTP packets. The CSRC list can have up to 15 members. There r in the CSRC list of the RTP packets. The CSRC list can have up to 15 members.
fore, text from up to 15 sources can be included in each packet. Packets are nor Therefore, text from up to 15 sources can be included in each packet. Packets ar
mally sent with 300 ms intervals. The mean delay will be 150 ms. A new redundanc e normally sent at 300 ms intervals. The mean delay would be 150 ms. A new redun
y packet format is specified. This method would result in good performance, but dancy packet format would be specified. This method would result in good perform
would require standardization and implementation of new releases in the target t ance but would require standardization and implementation of new releases in the
echnologies that would take more time than desirable to complete. It was therefo target technologies; these would take more time than desirable to complete. It
re not selected to be included in this document. was therefore not selected to be included in this document.
</dd> </dd>
<dt>Mixing for multiparty-unaware endpoints</dt> <dt>Mixing for multiparty-unaware endpoints:</dt>
<dd> <dd>
Presentation of text from multiple parties is prepared by the mixer in one singl The presentation of text from multiple parties is prepared by the mixer in one s
e stream. ingle stream.
It is desirable to have a method that does not require any modifications in ex It is desirable to have a method that does not require any modifications in ex
isting user devices implementing RFC 4103 for RTT without explicit support of mu isting user devices implementing RFC 4103 for real-time text without explicit su
ltiparty sessions. This is possible by having the mixer insert a new line and a pport of multiparty sessions. This is made possible by having the mixer insert a
text formatted source label before each switch of text source in the stream. Swi new line and a
tch of source can only be done in places in the text where it does not disturb t text-formatted source label before each switch of text source in the stream. Swi
he perception of the contents. Text from only one source can be presented in rea tching the source can only be done in places in the text where it does not distu
l time at a time. The delay will therefore vary. The method also has other limit rb the perception of the contents. Text from only one source at a time can be pr
ations, but is included in this document as a fallback method. In calls where pa esented in real time. The delay will therefore vary. In calls where parties take
rties take turns properly by ending their entries with a new line, the limitatio turns properly by ending their entries with a new line, the limitations will ha
ns will have limited influence on the user experience. when only two parties sen ve limited influence on the user experience. When only two parties send text, th
d text, these two will see the text in real time with no delay. This method is s ese two will see the text in real time with no delay. Although this method also
pecified as a fallback method in this document. has other limitations, it is included in this document as a fallback method.
</dd> </dd>
<dt> <dt>
RTT transport in WebRTC Real-time text transport in WebRTC:
</dt> </dt>
<dd> <dd>
Transport of real-time text in the WebRTC technology is specified to use the W ebRTC data channel in <xref target="RFC8865" format="default"/>. That specificat ion contains a section briefly describing its use in multiparty sessions. The fo cus of this document is RTP transport. Therefore, even if the WebRTC transport p rovides good multiparty performance, it is just mentioned in this document in re lation to providing gateways with multiparty capabilities between RTP and WebRTC technologies. <xref target="RFC8865" format="default"/> specifies how the WebRTC data channel can be used to transport real-time text. That specification contains a section b riefly describing its use in multiparty sessions. The focus of this document is RTP transport. Therefore, even if the WebRTC transport provides good multiparty performance, it is only mentioned in this document in relation to providing gate ways with multiparty capabilities between RTP and WebRTC technologies.
</dd> </dd>
</dl> </dl>
</section> </section>
<section anchor="intendedapp" numbered="true" toc="default"> <section anchor="intendedapp" numbered="true" toc="default">
<name>Intended application</name> <name>Intended Application</name>
<t> <t>
The method for multiparty real-time text specified in this document is p The method for multiparty real-time text specified in this document is p
rimarily intended for use in transmission between mixers and endpoints in centra rimarily intended for use in transmissions between mixers and endpoints in centr
lized mixing configurations. It is also applicable between mixers. An often ment alized mixing configurations. It is also applicable between mixers. An
ioned application is for emergency service calls with real-time text and voice, often-mentioned application is for emergency service calls with real-time text a
where a call taker wants to make an attended handover of a call to another agent nd voice, where a call taker wants to make an attended handover of a call to ano
, and stay to observe the session. Multimedia conference sessions with support f ther agent and stay on the call to observe the session. Multimedia conference se
or participants to contribute in text is another application. Conferences with c ssions with support for participants to contribute with text is another example.
entral support for speech-to-text conversion is yet another mentioned applicatio Conferences with central support for speech-to-text conversion represent yet a
n. nother example.
</t> </t>
<t> <t>
In all these applications, normally only one participant at a time wil l send long text utterances. In some cases, one other participant will occasiona lly contribute with a longer comment simultaneously. That may also happen in som e rare cases when text is interpreted to text in another language in a conferenc e. Apart from these cases, other participants are only expected to contribute wi th very brief utterings while others are sending text. In all these applications, normally only one participant at a time wil l send long text comments. In some cases, one other participant will occasionall y contribute with a longer comment simultaneously. That may also happen in some rare cases when text is translated to text in another language in a conference. Apart from these cases, other participants are only expected to contribute with very brief comments while others are sending text.
</t> </t>
<t> <t>
Users expect that the text they send is presented in real-time in Users expect the text they send to be presented in real time in
a readable way to the other participants even if they send simultaneously with a readable way to the other participants even if they send simultaneously with o
other users and even when they make brief edit operations of their text by backs ther users and even when they make brief edit operations of their text by backsp
pacing and correcting their text. acing and correcting their text.
</t> </t>
<t> <t>
Text is supposed to be human generated, by some text input means, such as typing on a keyboard or using speech-to-text technology. Occasional small cu t-and-paste operations may appear even if that is not the initial purpose of rea l-time text. Text is supposed to be human generated, by some means of text input, s uch as typing on a keyboard or using speech-to-text technology. Occasional small cut-and-paste operations may appear even if that is not the initial purpose of real-time text.
</t> </t>
<t> <t>
The real-time characteristics of real-time text is essential for the p articipants to be able to contribute to a conversation. If the text is too much delayed from typing a letter to its presentation, then, in some conference situa tions, the opportunity to comment will be gone and someone else will grab the tu rn. A delay of more than one second in such situations is an obstacle for good c onversation. The real-time characteristics of real-time text are essential for the participants to be able to contribute to a conversation. If the text is delayed too much between the typing of a character and its presentation, then, in some c onference situations, the opportunity to comment will be gone and someone else w ill grab the turn. A delay of more than one second in such situations is an obst acle to good conversation.
</t> </t>
</section> </section>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Overview of the two specified solutions and selection of method</name> <name>Overview of the Two Specified Solutions and Selection of Method</name>
<t> <t>
This section contains a brief introduction of the two methods specified in thi s document. This section contains a brief introduction of the two methods specified in thi s document.
</t> </t>
<section anchor="negouse" numbered="true" toc="default"> <section anchor="negouse" numbered="true" toc="default">
<name>The RTP-mixer-based solution for multiparty-aware endpoints</name> <name>The RTP-Mixer-Based Solution for Multiparty-Aware Endpoints</name>
<t> <t>
This method specifies negotiated use of the RFC 4103 format for multipart This method specifies the negotiated use of the formats described in RFC
y transmission in a single RTP stream. 4103, for multiparty transmissions in a single RTP stream.
The main purpose of this document is to specify a method for true multiparty rea The main purpose of this document is to specify a method for true multiparty rea
l-time text mixing for multiparty-aware endpoints that can be widely deployed. T l-time text mixing for multiparty-aware endpoints that can be widely deployed. T
he RTP-mixer-based method makes use of the current format for real-time text in he RTP-mixer-based method makes use of the current format for real-time text as
<xref target="RFC4103" format="default"/>. It is an update of RFC 4103 by a clar provided in <xref target="RFC4103" format="default"/>. This method updates RFC 4
ification on one way to use it in the multiparty situation. That is done by comp 103 by clarifying one way to use it in the multiparty situation. That is done by
leting a negotiation for this kind of multiparty capability and by interleaving completing a negotiation for this kind of multiparty capability and by interlea
packets from different sources. The source is indicated in the CSRC element in t ving packets from different sources. The source is indicated in the CSRC element
he RTP packets. Specific considerations are made to be able to recover text afte in the RTP packets. Specific considerations are made regarding the ability to r
r packet loss. ecover text after packet loss.
</t> </t>
<t> <t>
The detailed procedures for the RTP-mixer-based multiparty-aware case are specif ied in <xref target="nego2" format="default"/>. The detailed procedures for the RTP-mixer-based multiparty-aware case are specif ied in <xref target="nego2" format="default"/>.
</t> </t>
<t> <t>
Please use <xref target="RFC4103" format="default"/> as reference when readin g the specification. Please refer to <xref target="RFC4103" format="default"/> when reading this d ocument.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Mixing for multiparty-unaware endpoints</name> <name>Mixing for Multiparty-Unaware Endpoints</name>
<t> <t>
A method is also specified in this document for cases when the endpoint partic This document also specifies a method to be used in cases when the endpoint pa
ipating in a multiparty call does not itself implement any solution, or not the rticipating in a multiparty call does not itself implement any solution or does
same, as the mixer. The method requires the mixer to insert text dividers and re not implement the same solution as the mixer.
adable This method requires the mixer to insert text dividers and readable
labels and only send text from one source at a time until a suitable point a labels and only send text from one source at a time until a suitable point a
ppears for source change. ppears for changing the source.
This solution is a fallback method with functional limitations. It acts on t This solution is a fallback method with functional limitations. It operates
he presentation level. at the presentation level.
</t> </t>
<t> <t>
A mixer SHOULD by default format and transmit text to a call participant to be s A mixer <bcp14>SHOULD</bcp14> by default format and transmit text to a call
uitable to present on a multiparty-unaware endpoint which has not negotiated any participant so that the text is suitable for presentation on a
method for true multiparty RTT handling, but negotiated a "text/red" or "text/t multiparty-unaware endpoint that has not negotiated any method for
140" format in a session. This SHOULD be done if nothing else is specified for t true multiparty real-time text handling but has negotiated a "text/red" or
he application in order to maintain interoperability. <xref target="UnawareForma "text/t140" format in a session.
t" format="default"/> specifies how this mixing is done. This <bcp14>SHOULD</bcp14> be done if nothing else is specified for the applica
tion, in order to maintain interoperability. <xref target="UnawareFormat" format
="default"/> specifies how this mixing is done.
</t> </t>
</section> </section>
<section anchor="nego1" numbered="true" toc="default"> <section anchor="nego1" numbered="true" toc="default">
<name>Offer/answer considerations</name> <name>Offer/Answer Considerations</name>
<t> <t>
RTP Payload for Text Conversation <xref target="RFC4103" format="default"/> spec ifies use of RTP <xref target="RFC3550" format="default"/>, and a redundancy for mat "text/red" for increased robustness of real-time text transmission. "RTP Payload for Text Conversation" <xref target="RFC4103" format="default"/> sp ecifies the use of RTP <xref target="RFC3550" format="default"/> and a redundanc y format ("text/red", as defined in <xref target="RFC4102" format="default"/>) f or increased robustness of real-time text transmission.
This document updates <xref target="RFC4103" format="default"/> This document updates <xref target="RFC4103" format="default"/>
by introducing a capability negotiation for handling multiparty real-time text, a way to indicate the source of transmitted text, and rules for efficient timing of the transmissions interleaved from different sources. by introducing a capability negotiation for handling multiparty real-time text, a way to indicate the source of transmitted text, and rules for efficient timing of the transmissions interleaved from different sources.
</t> </t>
<t> <t>
The capability negotiation for the "RTP-mixer-based multiparty method" is based on use of the SDP media attribute "rtt-mixer". The capability negotiation for the RTP-mixer-based multiparty method is based on the use of the SDP media attribute "rtt-mixer".
</t> </t>
<dl newline="true"> <t>The syntax is as follows:</t>
<dt>The syntax is as follows:</dt> <sourcecode name="" type="sdp"><![CDATA[
<dd>"a=rtt-mixer"</dd> a=rtt-mixer
</dl> ]]></sourcecode>
<t> <t>
If any other method for RTP-based multiparty real-time text gets specified by ad ditional work, it is assumed that it will be recognized by some specific SDP fea ture exchange. If in the future any other method for RTP-based multiparty real-time text is spe cified by additional work, it is assumed that it will be recognized by some spec ific SDP feature exchange.
</t> </t>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Initial offer</name> <name>Initial Offer</name>
<t> <t>
A party intending to set up a session and being willing to use the RTP-mixer-bas ed method of this specification for sending or receiving or both sending and rec eiving real-time text SHALL include the "rtt-mixer" SDP attribute in the corresp onding "text" media section in the initial offer. A party that intends to set up a session and is willing to use the RTP-mixer-bas ed method provided in this specification for sending, receiving, or both sending and receiving real-time text <bcp14>SHALL</bcp14> include the "rtt-mixer" SDP a ttribute in the corresponding "text" media section in the initial offer.
</t> </t>
<t> <t>
The party MAY indicate capability for both the RTP-mixer-based method of this sp ecification and other methods. The party <bcp14>MAY</bcp14> indicate its capability regarding both the RTP-mixe r-based method provided in this specification and other methods.
</t> </t>
<t> <t>
When the offeror has sent the offer including the "rtt-mixer" attribute, it MUST be prepared to receive and handle real-time text formatted according to both th e method for multiparty-aware parties specified in <xref target="nego2" format=" default"/> in this specification and two-party formatted real-time text. When the offerer has sent the offer, which includes the "rtt-mixer" attribute, i t <bcp14>MUST</bcp14> be prepared to receive and handle real-time text formatted according to both the method for multiparty-aware parties specified in <xref ta rget="nego2" format="default"/> and two-party formatted real-time text.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Answering the offer</name> <name>Answering the Offer</name>
<t> <t>
A party receiving an offer containing the "rtt-mixer" SDP attribute and being willing to use the RTP-mixer-based method of this specification for sending or r eceiving or both sending and receiving SHALL include the "rtt-mixer" SDP attribu te in the corresponding "text" media section in the answer. A party that receives an offer containing the "rtt-mixer" SDP attribute and is willing to use the RTP-mixer-based method provided in this specification for se nding, receiving, or both sending and receiving real-time text <bcp14>SHALL</bcp 14> include the "rtt-mixer" SDP attribute in the corresponding "text" media sect ion in the answer.
</t> </t>
<t> <t>
If the offer did not contain the "rtt-mixer" attribute, the answer MUST NOT co ntain the "rtt-mixer" attribute. If the offer did not contain the "rtt-mixer" attribute, the answer <bcp14>MUST NOT</bcp14> contain the "rtt-mixer" attribute.
</t> </t>
<t> <t>
Even when the "rtt-mixer" attribute is successfully negotiated, the parties MA Y send and receive two-party coded real-time text. Even when the "rtt-mixer" attribute is successfully negotiated, the parties <b cp14>MAY</bcp14> send and receive two-party coded real-time text.
</t> </t>
<t> <t>
An answer MUST NOT include acceptance of more than one method for multiparty r eal-time text in the same RTP session. An answer <bcp14>MUST NOT</bcp14> include acceptance of more than one method f or multiparty real-time text in the same RTP session.
</t> </t>
<t> <t>
When the answer including acceptance is transmitted, the answerer MUST be prep When the answer, which includes acceptance, is transmitted, the answerer <bcp1
ared to act on received text in the negotiated session according to the method f 4>MUST</bcp14> be prepared to act on received text in the negotiated session acc
or multiparty-aware parties specified in <xref target="nego2" format="default"/> ording to the method for multiparty-aware parties specified in <xref target="neg
of this specification. Reception of text for a two-party session SHALL also be o2" format="default"/>. Reception of text for a two-party session <bcp14>SHALL</
supported. bcp14> also be supported.
</t>
<t>
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Offeror processing the answer</name> <name>Offerer Processing the Answer</name>
<t> <t>
When the answer is processed by the offeror, it MUST act as specified in <xref target="negoresult" format="default"/> When the answer is processed by the offerer, the offerer <bcp14>MUST</bc p14> follow the requirements listed in <xref target="negoresult" format="default "/>.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Modifying a session</name> <name>Modifying a Session</name>
<t> <t>
A session MAY be modified at any time by any party offering a modified SD P with or without the "rtt-mixer" SDP attribute expressing a desired change in t he support of multiparty real-time text. A session <bcp14>MAY</bcp14> be modified at any time by any party offeri ng a modified SDP with or without the "rtt-mixer" SDP attribute expressing a des ired change in the support of multiparty real-time text.
</t> </t>
<t> <t>
If the modified offer adds indication of support for multiparty real-time text by including the "rtt-mixer" SDP attribute, the procedures specified in the pre vious subsections SHALL be applied. If the modified offer adds the indication of support for multiparty real-time text by including the "rtt-mixer" SDP attribute, the procedures specified in the previous subsections <bcp14>SHALL</bcp14> be applied.
</t> </t>
<t> <t>
If the modified offer deletes indication of support for multiparty real-time t ext by excluding the "rtt-mixer" SDP attribute, the answer MUST NOT contain the "rtt-mixer" attribute. After processing this SDP exchange, the parties MUST NOT send real-time text formatted for multiparty-aware parties according to this spe cification. If the modified offer deletes the indication of support for multiparty real-ti me text by excluding the "rtt-mixer" SDP attribute, the answer <bcp14>MUST NOT</ bcp14> contain the "rtt-mixer" attribute. After processing this SDP exchange, th e parties <bcp14>MUST NOT</bcp14> send real-time text formatted for multiparty-a ware parties according to this specification.
</t> </t>
</section> </section>
</section> </section>
<section anchor="negoresult" numbered="true" toc="default"> <section anchor="negoresult" numbered="true" toc="default">
<name>Actions depending on capability negotiation result</name> <name>Actions Depending on Capability Negotiation Result</name>
<t> <t>
A transmitting party SHALL send text according to the RTP-mixer-based multipar ty method only when the negotiation for that method was successful and when it c onveys text for another source. In all other cases, the packets SHALL be populat ed and interpreted as for a two-party session. A transmitting party <bcp14>SHALL</bcp14> send text according to the RTP-mixer -based multiparty method only when the negotiation for that method was successfu l and when it conveys text for another source. In all other cases, the packets < bcp14>SHALL</bcp14> be populated and interpreted as for a two-party session.
</t> </t>
<t> <t>
A party which has negotiated the "rtt-mixer" SDP media attribute MUST populate the CSRC-list, and format the packets according to <xref target="nego2" format= "default"/> if it acts as an rtp-mixer and sends multiparty text. A party that has negotiated the "rtt-mixer" SDP media attribute and acts as an RTP mixer sending multiparty text <bcp14>MUST</bcp14> (1) populate the CSRC lis t and (2) format the packets according to <xref target="nego2" format="default"/ >.
</t> </t>
<t> <t>
A party which has negotiated the "rtt-mixer" SDP media attribute MUST interpre t the contents of the "CC" field, the CSRC-list and the packets according to <xr ef target="nego2" format="default"/> in received RTP packets in the correspondin g RTP stream. A party that has negotiated the "rtt-mixer" SDP media attribute <bcp14>MUST</b cp14> interpret the contents of the CC field, the CSRC list, and the packets acc ording to <xref target="nego2" format="default"/> in received RTP packets in the corresponding RTP stream.
</t> </t>
<t> <t>
A party which has not successfully completed the negotiation of the "rt A party that has not successfully completed the negotiation of the "rt
t-mixer" SDP media attribute MUST NOT transmit packets interleaved from differen t-mixer" SDP media attribute <bcp14>MUST NOT</bcp14> transmit packets interleave
t sources in the same RTP stream as specified in <xref target="nego2" format="de d from different sources in the same RTP stream, as specified in <xref target="n
fault"/>. ego2" format="default"/>.
If the party is a mixer and did declare the "rtt-mixer" SDP media attri If the party is a mixer and did declare the "rtt-mixer" SDP media attr
bute, it SHOULD perform the procedure for multiparty-unaware endpoints. If the p ibute, it <bcp14>SHOULD</bcp14> perform the procedure for multiparty-unaware end
arty is not a mixer, it SHOULD transmit as in a two-party session according to < points. If the party is not a mixer, it <bcp14>SHOULD</bcp14> transmit as in a t
xref target="RFC4103" format="default"/>. wo-party session according to <xref target="RFC4103" format="default"/>.
</t> </t>
</section> </section>
</section> </section>
<section anchor="nego2" numbered="true" toc="default"> <section anchor="nego2" numbered="true" toc="default">
<name>Details for the RTP-mixer-based mixing method for multiparty-aware endpo ints</name> <name>Details for the RTP-Mixer-Based Mixing Method for Multiparty-Aware Endpo ints</name>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Use of fields in the RTP packets</name> <name>Use of Fields in the RTP Packets</name>
<t> <t>
The CC field SHALL show the number of members in the CSRC list, which SHALL be o ne (1) in transmissions from a mixer when conveying text from other sources in a multiparty session, and otherwise 0. The CC field <bcp14>SHALL</bcp14> show the number of members in the CSRC list, w hich <bcp14>SHALL</bcp14> be one (1) in transmissions from a mixer when conveyi ng text from other sources in a multiparty session, and otherwise 0.
</t> </t>
<t> <t>
When text is conveyed by a mixer during a multiparty session, a CSRC list SHALL be included in the packet. The single member in the CSRC-list SHALL contain the SSRC of the source of the T140blocks in the packet. When text is conveyed by a mixer during a multiparty session, a CSRC list <bcp14 >SHALL</bcp14> be included in the packet. The single member in the CSRC list <bc p14>SHALL</bcp14> contain the SSRC of the source of the T140blocks in the packet .
</t> </t>
<t> <t>
When redundancy is used, the RECOMMENDED level of redundancy is to use one prim ary and two redundant generations of T140blocks. In some cases, a primary or red undant T140block is empty, but is still represented by a member in the redundanc y header. When redundancy is used, the <bcp14>RECOMMENDED</bcp14> level of redundancy is to use one primary and two redundant generations of T140blocks. In some cases, a primary or redundant T140block is empty but is still represented by a member in the redundancy header.
</t> </t>
<t> <t>
In other regards, the contents of the RTP packets are equal to what is specifi In other respects, the contents of the RTP packets will be as specified in <xr
ed in <xref target="RFC4103" format="default"/>. ef target="RFC4103" format="default"/>.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Initial transmission of a BOM character</name> <name>Initial Transmission of a BOM Character</name>
<t> <t>
As soon as a participant is known to participate in a session with another As soon as a participant is known to participate in a session with another
entity and is available for text reception, a Unicode Byte-Order Mark (BOM) cha entity and is available for text reception, a Unicode byte order mark (BOM) cha
racter SHALL be sent to it by the other entity according to the procedures in th racter <bcp14>SHALL</bcp14> be sent to it by the other entity according to the p
is section. This is useful in many configurations to open ports and firewalls an rocedures in this section. This is useful in many configurations for opening por
d setting up the connection between the application and the network. If the tran ts and firewalls and for setting up the connection between the application and t
smitter is a mixer, then the source of this character SHALL be indicated to be t he network. If the transmitter is a mixer, then the source of this character <bc
he mixer itself. p14>SHALL</bcp14> be indicated to be the mixer itself.
</t> </t>
<t> <t>
Note that the BOM character SHALL be transmitted with the same red undancy procedures as any other text. Note that the BOM character <bcp14>SHALL</bcp14> be transmitted wi th the same redundancy procedures as any other text.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Keep-alive</name> <section numbered="true" toc="default"><name>Keep-Alive</name>
<t> <t>
After that, the transmitter SHALL send keep-alive traffic to the receiver( After that, the transmitter <bcp14>SHALL</bcp14> send keep-alive traffic t
s) at regular intervals when no other traffic has occurred during that interval, o the receiver(s) at regular intervals when no other traffic has occurred during
if that is decided for the actual connection. It is RECOMMENDED to use the keep that interval, if that is decided upon for the actual connection. It is <bcp14>
-alive solution from <xref target="RFC6263" format="default"/>. The consent chec RECOMMENDED</bcp14> to use the keep-alive solution provided in <xref target="RFC
k of <xref target="RFC7675" format="default"/> is a possible alternative if it i 6263" format="default"/>. The consent check <xref target="RFC7675" format="defau
s used anyway for other reasons. lt"/> is a possible alternative if it is used anyway for other reasons.
</t> </t>
</section> </section>
<section numbered="true" anchor="transmint" toc="default"><name>Transm ission interval</name> <section numbered="true" anchor="transmint" toc="default"><name>Transm ission Interval</name>
<t> <t>
A "text/red" or "text/t140" transmitter in a mixer SHALL send packets dist A "text/red" or "text/t140" transmitter in a mixer <bcp14>SHALL</bcp14> se
ributed in time as long as there is something (new or redundant T140blocks) to t nd packets distributed over time as long as there is something (new or redundant
ransmit. The maximum transmission interval between text transmissions from the T140blocks) to transmit.
same source SHALL then be 330 ms, when no other limitations cause a longer inter The maximum transmission interval between text transmissions from the same sou
val to be temporarily used. It is RECOMMENDED to send the next packet to a recei rce <bcp14>SHALL</bcp14> then be 330 ms, when no other limitations cause a longe
ver as soon as new text to that receiver is available, as long as the mean chara r interval to be temporarily used. It is <bcp14>RECOMMENDED</bcp14> to send the
cter rate of new text to the receiver calculated over the last 10 one-second int next packet to a receiver as soon as new text to that receiver is available, as
ervals does not exceed the "cps" value of the receiver. The intention is to keep long as the mean character rate of new text to the receiver calculated over the
the latency low and network load limited while keeping good protection against last 10 one-second intervals does not exceed the "cps" value of the receiver. Th
text loss in bursty packet loss conditions. The main purpose of the 330 ms inter e intention is to keep the latency low and network load limited while keeping go
val is for timing of redundant transmission, when no new text from the same sour od protection against text loss in bursty packet loss conditions. The main purpo
ce is available. se of the 330 ms interval is for the timing of redundant transmissions, when no
new text from the same source is available.
</t> </t>
<t> <t>
The reason for the value 330 ms is that many sources of text will The value of 330 ms is used, because many sources of text will
transmit new text with 300 ms intervals during periods of transmit new text at 300 ms intervals during periods of
continuous user typing, and then reception in the mixer of such new continuous user typing, and then reception in the mixer of such new
text will cause a combined transmission of the new text and the text will cause a combined transmission of the new text and the
unsent redundancy from the previous transmission. Only when the user unsent redundancy from the previous transmission. Only when the user
stops typing, the 330 ms interval will be applied to send the stops typing will the 330 ms interval be applied to send the
redundancy. redundancy.
</t> </t>
<t> <t>
If the Characters Per Second (cps) value is reached, a longer transmissio If the characters per second ("cps") value is reached, a longer transmis
n interval SHALL be applied for text from all sources as specified in <xref tar sion interval <bcp14>SHALL</bcp14> be applied for text from all sources as speci
get="RFC4103" format="default"/> and only as much of the text queued for transmi fied in <xref target="RFC4103" format="default"/> and only as much of the text
ssion SHALL be sent at the end of each transmission interval as can be allowed w queued for transmission <bcp14>SHALL</bcp14> be sent at the end of each transmis
ithout exceeding the "cps" value. Division of text for partial transmission MUST sion interval as can be allowed without exceeding the "cps" value. Division of t
then be made at T140block borders. When the transmission rate falls under the " ext for partial transmission <bcp14>MUST</bcp14> then be made at T140block borde
cps" value again, rs. When the transmission rate falls below the "cps" value again,
the transmission intervals SHALL be returned to 330 ms and transmission of ne the transmission intervals <bcp14>SHALL</bcp14> be reset to 330 ms and transm
w text SHALL return to be made as soon as new ission of new text <bcp14>SHALL</bcp14> again be made as soon as new
text is available. text is available.
</t> </t>
<t>
NOTE: that extending the transmission intervals during high load <aside><t>NOTE: Extending the transmission intervals during periods of high l
periods does not change the number of characters to be conveyed. oad
It just evens out the load in time and reduces the number of packets does not change the number of characters to be conveyed.
per second. With human created conversational text, the sending It just evens out the load over time and reduces the number of packets
user will eventually take a pause letting transmission catch up. per second. With human-created conversational text, the sending
</t> user will eventually take a pause, letting transmission catch up.</t></aside>
<t> <t>
See also <xref target="congest" format="default"/>. See also <xref target="congest" format="default"/>.
</t> </t>
<t> <t>
For a transmitter not acting as a mixer, the transmission interval princi For a transmitter not acting as a mixer, the transmission interval princ
ples from <xref target="RFC4103" format="default"/> apply, and the normal trans iples provided in <xref target="RFC4103" format="default"/> apply, and the norma
mission interval SHALL be 300 ms. l transmission interval <bcp14>SHALL</bcp14> be 300 ms.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Only one source per packet</name> <section numbered="true" toc="default"><name>Only One Source per Packet</name>
<t> New text and redundant copies of earlier text from one source SHALL be t <t> New text and redundant copies of earlier text from one source <bcp14>SHA
ransmitted in the same packet if available for transmission at the same time. Te LL</bcp14> be transmitted in the same packet if available for transmission at th
xt from different sources MUST NOT be transmitted in the same packet. e same time. Text from different sources <bcp14>MUST NOT</bcp14> be transmitted
</t> in the same packet.
</t>
</section> </section>
<section numbered="true" toc="default"><name>Do not send received text to the originating source</name> <section numbered="true" toc="default"><name>Do Not Send Received Text to the Originating Source</name>
<t> <t>
Text received by a mixer from a participant SHOULD NOT be included Text received by a mixer from a participant <bcp14>SHOULD NOT</bcp
in transmission from the mixer to that participant, because the normal behavior 14> be included in transmissions from the mixer to that participant, because for
of the endpoint is to present locally-produced text locally. text
that is produced locally, the normal behavior of the endpoint is to
present such text directly when it is produced.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Clean incoming text</na me> <section numbered="true" toc="default"><name>Clean Incoming Text</na me>
<t> <t>
A mixer SHALL handle reception, recovery from packet loss, deletio n of superfluous redundancy, marking of possible text loss and deletion of 'BOM' characters from each participant before queueing received text for transmission to receiving participants as specified in <xref target="RFC4103" format="defaul t"/> for single-party sources and <xref target="reception" format="default"/> fo r multiparty sources (chained mixers). A mixer <bcp14>SHALL</bcp14> handle reception, recovery from packe t loss, deletion of superfluous redundancy, marking of possible text loss, and d eletion of BOM characters from each participant before queueing received text fo r transmission to receiving participants as specified in <xref target="RFC4103" format="default"/> for single-party sources and <xref target="reception" format ="default"/> for multiparty sources (chained mixers).
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Redundant transmission pr <section numbered="true" toc="default"><name>Principles of Redundant T
inciples</name> ransmission</name>
<t> <t>
A transmitting party using redundancy SHALL send redundant repetitions of A transmitting party using redundancy <bcp14>SHALL</bcp14> send redundant
T140blocks already transmitted in earlier packets. repetitions of T140blocks already transmitted in earlier packets.
</t> </t>
<t> The number of redundant generations of T140blocks to include in transmit <t> The number of redundant generations of T140blocks to include in transmit
ted packets SHALL be deduced from the SDP negotiation. It SHALL be set to the mi ted packets <bcp14>SHALL</bcp14> be deduced from the SDP negotiation. It <bcp14>
nimum of the number declared by the two parties negotiating a connection. It is SHALL</bcp14> be set to the minimum of the number declared by the two parties ne
RECOMMENDED to declare and transmit one original and two redundant generations o gotiating a connection. It is <bcp14>RECOMMENDED</bcp14> to declare and transmit
f the T140blocks because that provides good protection against text loss in case one original and two redundant generations of the T140blocks, because this prov
of packet loss, and low overhead. ides good protection against text loss in the case of packet loss and also provi
</t> des low overhead.
</t>
</section> </section>
<section numbered="true" toc="default"><name>Text placement in packets <section numbered="true" toc="default"><name>Text Placement in Packets
</name> </name>
<t> <t>
The mixer SHALL compose and transmit an RTP packet to a receive The mixer <bcp14>SHALL</bcp14> compose and transmit an RTP pac
r when one or more of the following conditions have occurred:</t> ket to a receiver when one or more of the following conditions have occurred:</t
<ul> >
<li>The transmission interval is the normal 330 ms and th <ul>
ere is newly received unsent text available for transmission to that receiver.</ <li>The transmission interval is the normal 330 ms (no m
li> atter whether the transmission interval has passed or not), and there is newly r
<li>The current transmission interval has passed and is l eceived unsent text available for transmission to that receiver.</li>
onger than the normal 330 ms and there is newly received unsent text available f <li>The current transmission interval has passed and is
or transmission to that receiver.</li> longer than the normal 330 ms, and there is newly received unsent text available
<li>The current transmission interval ( normally 330 ms) for transmission to that receiver.</li>
has passed since already transmitted text was queued for transmission as redunda <li>The current transmission interval (normally 330 ms)
nt text.</li> has passed since already-transmitted text was queued for transmission as redunda
</ul> nt text.</li>
</ul>
<t> <t>
The principles from <xref target="RFC4103" format="defaul The principles provided in <xref target="RFC4103" format
t"/> apply for populating the header, the redundancy header and the ="default"/> apply for populating the header, the redundancy header, and the
data in the packet with specifics specified here and in the following sections. data in the packet with specific information, as detailed here and in the follow
ing sections.
</t> </t>
<t> <t>
At the time of transmission, the mixer SHALL populate the RTP packe At the time of transmission, the mixer <bcp14>SHALL</bcp14>
t with all T140blocks queued for transmission originating from the source in tur populate the RTP packet with all T140blocks queued for transmission originating
n for transmission as long as this is not in conflict with the allowed number of from the source selected for transmission as long as this is not in conflict wit
characters per second ("cps") or the maximum packet size. In this way, the late h the allowed number of characters per second ("cps") or the maximum packet size
ncy of the latest received text is kept low even in moments of simultaneous tran .
smission from many sources. In this way, the latency of the latest received text is kept low even in moment
</t> s of simultaneous transmission from many sources.
<t>
Redundant text SHALL also be included, and the assessment of how much n
ew text can be included within the maximum packet size MUST take into account th
at the redundancy has priority to be transmitted in its entirety. See <xref tar
get="transmint" format="default"/>
</t>
<t>
The SSRC of the source SHALL be placed as the only member in the CSRC-l
ist.
</t>
<t>
Note: The CSRC-list in an RTP packet only includes the participant whose text is
included in text blocks. It is not the same as the total list of participants i
n a conference. With audio and video media, the CSRC-list would often contain al
l participants who are not muted whereas text participants that don't type are c
ompletely silent and thus are not represented in RTP packet CSRC-lists.
</t> </t>
<t>
Redundant text <bcp14>SHALL</bcp14> also be included, and the assessme
nt of how much new text can be included within the maximum packet size <bcp14>MU
ST</bcp14> take into account that the redundancy has priority to be transmitted
in its entirety. See <xref target="transmint" format="default"/>.
</t>
<t>
The SSRC of the source <bcp14>SHALL</bcp14> be placed as the only memb
er in the CSRC list.
</t>
<aside><t>
Note: The CSRC list in an RTP packet only includes the participant whose text is
included in text blocks. It is not the same as the total list of participants i
n a conference. With audio and video media, the CSRC list would often contain al
l participants who are not muted, whereas text participants that don't type are
completely silent and thus are not represented in RTP packet CSRC lists.
</t></aside>
</section> </section>
<section numbered="true" toc="default"><name>Empty T140blocks</name> <section numbered="true" toc="default"><name>Empty T140blocks</name>
<t> <t>
If no unsent T140blocks were available for a source at the time of populat ing a packet, but T140blocks are available which have not yet been sent the full intended number of redundant transmissions, then the primary T140block for that source is composed of an empty T140block, and populated (without taking up any length) in a packet for transmission. The corresponding SSRC SHALL be placed as usual in its place in the CSRC-list. If no unsent T140blocks were available for a source at the time of populat ing a packet but already-transmitted T140blocks are available that have not yet been sent the full intended number of redundant transmissions, then the primary area in the packet is composed of an empty T140block and included (without takin g up any length) in the packet for transmission. The corresponding SSRC <bcp14>S HALL</bcp14> be placed as usual in its place in the CSRC list.
</t> </t>
<t> <t>
The first packet in the session, the first after a source switch, and th The first packet in the session, the first after a source switch, and th
e first after a pause SHALL be populated with the e first after a pause <bcp14>SHALL</bcp14> be populated with the
available T140blocks for the source in turn to be sent as primary, and e available T140blocks for the source selected to be sent as the primary,
mpty T140blocks for the agreed number of redundancy generations. and empty T140blocks for the agreed-upon number of redundancy generations.
</t> </t>
</section> </section>
<section anchor="redundancy" numbered="true" toc="default"><name>Creation of the redundancy</name> <section anchor="redundancy" numbered="true" toc="default"><name>Creation of the Redundancy</name>
<t> <t>
The primary T140block from a source in the latest transmitted packet is sa ved for populating the first redundant T140block for that source in the next tra nsmission of text from that source. The first redundant T140block for that sourc e from the latest transmission is saved for populating the second redundant T140 block in the next transmission of text from that source. The primary T140block from a source in the latest transmitted packet is sa ved for populating the first redundant T140block for that source in the next tra nsmission of text from that source. The first redundant T140block for that sourc e from the latest transmission is saved for populating the second redundant T140 block in the next transmission of text from that source.
</t> </t>
<t> <t>
Usually this is the level of redundancy used. If a higher level of redunda ncy is negotiated, then the procedure SHALL be maintained until all available re dundant levels of T140blocks are placed in the packet. If a receiver has negotia ted a lower number of "text/red" generations, then that level SHALL be the maxim um used by the transmitter. Usually, this is the level of redundancy used. If a higher level of redund ancy is negotiated, then the procedure <bcp14>SHALL</bcp14> be continued until a ll available redundant levels of T140blocks are placed in the packet. If a recei ver has negotiated a lower number of "text/red" generations, then that level <bc p14>SHALL</bcp14> be the maximum used by the transmitter.
</t> </t>
<t> <t>
The T140blocks saved for transmission as redundant data are assigned a The T140blocks saved for transmission as redundant data are assigned a
planned transmission time 330 ms after the current time, but SHOULD be transmitt planned transmission time of 330 ms after the current time but <bcp14>SHOULD</b
ed earlier if new text for the same source gets in turn for transmission before cp14> be transmitted earlier if new text for the same source gets selected for t
that time. ransmission before that time.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Timer offset fields</name> <section numbered="true" toc="default"><name>Timer Offset Fields</name>
<t> <t>
The timestamp offset values SHALL be inserted in the redundancy header, wi th the time offset from the RTP timestamp in the packet when the corresponding T 140block was sent as primary. The timestamp offset values <bcp14>SHALL</bcp14> be inserted in the redund ancy header, with the time offset from the RTP timestamp in the packet when the corresponding T140block was sent as the primary.
</t> </t>
<t> <t>
The timestamp offsets are expressed in the same clock tick units as the RTP timestamp. The timestamp offsets are expressed in the same clock tick units as the RTP timestamp.
</t> </t>
<t> <t>
The timestamp offset values for empty T140blocks have no relevance but S HOULD be assigned realistic values. The timestamp offset values for empty T140blocks have no relevance but < bcp14>SHOULD</bcp14> be assigned realistic values.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Other RTP header fields</name > <section numbered="true" toc="default"><name>Other RTP Header Fields</name >
<t> <t>
The number of members in the CSRC list (0 or 1) SHALL be placed in the "CC The number of members in the CSRC list (0 or 1) <bcp14>SHALL</bcp14> be pl
" header field. Only mixers place value 1 in the "CC" field. A value of "0" indi aced in the CC header field. Only mixers place value 1 in the CC field. A value
cates that the source is the transmitting device itself and that the source is i of 0 indicates that the source is the transmitting device itself and that the so
ndicated by the SSRC field. This value is used by endpoints, and by mixers sendi urce is indicated by the SSRC field. This value is used by endpoints and also by
ng self-sourced data. mixers sending self-sourced data.
</t> </t>
<t> <t>
The current time SHALL be inserted in the timestamp. The current time <bcp14>SHALL</bcp14> be inserted in the timesta mp.
</t> </t>
<t> <t>
The SSRC header field SHALL contain the SSRC of the RTP session where th e packet will be transmitted. The SSRC header field <bcp14>SHALL</bcp14> contain the SSRC of the RTP s ession where the packet will be transmitted.
</t> </t>
<t> <t>
The M-bit SHALL be handled as specified in <xref target="RFC4103" format ="default"/>. The M-bit <bcp14>SHALL</bcp14> be handled as specified in <xref target=" RFC4103" format="default"/>.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Pause in transmission</name> <section numbered="true" toc="default"><name>Pause in Transmission</name>
<t> <t>
When there is no new T140block to transmit, and no redundant T140block tha t has not been retransmitted the intended number of times from any source, the t ransmission process SHALL be stopped until either new T140blocks arrive, or a ke ep-alive method calls for transmission of keep-alive packets. When there is no new T140block to transmit and no redundant T140block that has not been retransmitted the intended number of times from any source, the tr ansmission process <bcp14>SHALL</bcp14> be stopped until either new T140blocks a rrive or a keep-alive method calls for transmission of keep-alive packets.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>RTCP considerations</name> <name>RTCP Considerations</name>
<t> <t>
A mixer SHALL send RTCP reports with SDES, CNAME, and NAME information abo ut the sources in the multiparty call. This makes it possible for participants t o compose a suitable label for text from each source. A mixer <bcp14>SHALL</bcp14> send RTCP reports with SDES, CNAME, and NAME information about the sources in the multiparty call. This makes it possible for participants to compose a suitable label for text from each source.
</t> </t>
<t> <t>
Privacy considerations SHALL be taken when composing these fields. Privacy considerations <bcp14>SHALL</bcp14> be taken when composin
They contain name and address information that may be sensitive to transmit in g these fields. They contain name and address information that may be considered
its entirety, e.g., to unauthenticated participants. sensitive if the information is transmitted in its entirety, e.g., to unauthenti
cated participants.
</t> </t>
</section> </section>
<section anchor="reception" numbered="true" toc="default"> <section anchor="reception" numbered="true" toc="default">
<name>Reception of multiparty contents</name> <name>Reception of Multiparty Contents</name>
<t> <t>
The "text/red" receiver included in an endpoint with presentation functions will The "text/red" receiver included in an endpoint with presentation functions will
receive RTP packets in the single stream from the mixer, and SHALL distribute t receive RTP packets in the single stream from the mixer and <bcp14>SHALL</bcp14
he T140blocks for presentation in presentation areas for each source. Other rece > distribute the T140blocks for presentation in presentation areas for each sour
iver roles, such as gateways or chained mixers, are also feasible. They require ce. Other receiver roles, such as gateways or chained mixers, are also feasible.
considerations if the stream shall just be forwarded, or distributed based on th Whether the stream will only be forwarded or will be distributed based on the di
e different sources. fferent sources must be taken into consideration.
</t> </t>
<section numbered="true" toc="default"><name>Acting on the source of the packet contents</name> <section numbered="true" toc="default"><name>Acting on the Source of the Packet Contents</name>
<t> <t>
If the "CC" field value of a received packet is 1, it indicates that If the CC field value of a received packet is 1, it indicates that
the text is conveyed from a source indicated in the single member in the CSRC the text is conveyed from a source indicated in the single member in the CSRC
-list, and the receiver MUST act on the source according to its role. If the CC list, and the receiver <bcp14>MUST</bcp14> act on the source according to its r
value is 0, the source is indicated in the SSRC field. ole. If the CC value is 0, the source is indicated in the SSRC field.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Detection and indication of possibl e text loss</name> <section numbered="true" toc="default"><name>Detection and Indication of Possibl e Text Loss</name>
<t> <t>
The receiver SHALL monitor the RTP sequence numbers of the received packets for The receiver <bcp14>SHALL</bcp14> monitor the RTP sequence numbers of the receiv
gaps and packets out of order. ed packets for gaps and for packets received out of order.
If a sequence number gap appears and still exists after some defined short time If a sequence number gap appears and still exists after some defined short time
for jitter and reordering resolution, the packets in the gap SHALL be regarded a for jitter and reordering resolution, the packets in the gap <bcp14>SHALL</bcp14
s lost. > be regarded as lost.
</t> </t>
<t> <t>
If it is known that only one source is active in the RTP session, then it is lik ely that a gap equal to or larger than the agreed number of redundancy generatio ns (including the primary) causes text loss. In that case, the receiver SHALL cr eate a t140block with a marker for possible text loss <xref target="T140ad1" for mat="default"/> and associate it with the source and insert it in the reception buffer for that source. If it is known that only one source is active in the RTP session, then it is lik ely that a gap equal to or larger than the agreed-upon number of redundancy gene rations (including the primary) causes text loss. In that case, the receiver <bc p14>SHALL</bcp14> create a T140block with a marker for possible text loss <xref target="T140ad1" format="default"/>, associate it with the source, and insert it in the reception buffer for that source.
</t> </t>
<t> <t>
If it is known that more than one source is active in the RTP session, then it i s not possible in general to evaluate if text was lost when packets were lost. W ith two active sources and the recommended number of redundancy generations (3), it can take a gap of five consecutive lost packets until any text may be lost, but text loss can also appear if three non-consecutive packets are lost when the y contained consecutive data from the same source. A simple method to decide whe n there is risk for resulting text loss is to evaluate if three or more packets were lost within one second. If this simple method is used, then a t140block SHO ULD be created with a marker for possible text loss <xref target="T140ad1" forma t="default"/> and associated with the SSRC of the RTP session as a general input from the mixer. If it is known that more than one source is active in the RTP session, then it i s not possible in general to evaluate if text was lost when packets were lost. W ith two active sources and the recommended number of redundancy generations (one original and two redundant), it can take a gap of five consecutive lost packets before any text may be lost, but text loss can also appear if three non-consecu tive packets are lost when they contained consecutive data from the same source. A simple method for deciding when there is a risk of resulting text loss is to evaluate if three or more packets were lost within one second. If this simple me thod is used, then a T140block <bcp14>SHOULD</bcp14> be created with a marker fo r possible text loss <xref target="T140ad1" format="default"/> and associated wi th the SSRC of the RTP session as a general input from the mixer.
</t> </t>
<t> <t>
Implementations MAY apply more refined methods for more reliable detection of wh ether text was lost or not. Any refined method SHOULD prefer marking possible lo ss rather than not marking when it is uncertain if there was loss. Implementations <bcp14>MAY</bcp14> apply more refined methods for more reliable detection of whether text was lost or not. Any refined method <bcp14>SHOULD</bcp 14> prefer marking possible loss rather than not marking when it is uncertain if there was loss.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Extracting text and handling recove ry</name> <section numbered="true" toc="default"><name>Extracting Text and Handling Recove ry</name>
<t> <t>
When applying the following procedures, the effects MUST be considered of possib le timestamp wrap around and the RTP session possibly changing SSRC. When applying the following procedures, the effects of possible timestamp wrapar ound and the RTP session possibly changing the SSRC <bcp14>MUST</bcp14> be consi dered.
</t> </t>
<t> <t>
When a packet is received in an RTP session using the packetization for multipar ty-aware endpoints, its T140blocks SHALL be extracted in the following way. When a packet is received in an RTP session using the packetization for multipar ty-aware endpoints, its T140blocks <bcp14>SHALL</bcp14> be extracted as describe d below.
</t> </t>
<t> <t>
The source SHALL be extracted from the CSRC-list if available, otherwise from th e SSRC. The source <bcp14>SHALL</bcp14> be extracted from the CSRC list if available, an d otherwise from the SSRC.
</t> </t>
<t> <t>
If the received packet is the first packet received from the source, then all T1 40blocks in the packet SHALL be retrieved and assigned to a receive buffer for t he source beginning with the oldest available redundant generation, continuing w ith the younger redundant generations in age order and finally the primary. If the received packet is the first packet received from the source, then all T1 40blocks in the packet <bcp14>SHALL</bcp14> be retrieved and assigned to a recei ve buffer for that source, beginning with the oldest available redundant generat ion, continuing with the younger redundant generations in age order, and finally ending with the primary.
</t> </t>
<t> <aside><t>
Note: The normal case is that in the first packet, only the primary data has con tents. The redundant data has contents in the first received packet from a sourc e only after initial packet loss. Note: The normal case is that in the first packet, only the primary data has con tents. The redundant data has contents in the first received packet from a sourc e only after initial packet loss.
</t> </t></aside>
<t> <t>
If the packet is not the first packet from a source, then if redundant If the packet is not the first packet from a source, then if redundant
data is available, the process SHALL start with the oldest generation. data is available, the process <bcp14>SHALL</bcp14> start with the oldest gen
The timestamp of that redundant data SHALL be eration.
The timestamp of that redundant data <bcp14>SHALL</bcp14> be
created by subtracting its timestamp offset from the RTP timestamp. created by subtracting its timestamp offset from the RTP timestamp.
If the resulting timestamp is later than the latest retrieved data If the resulting timestamp is later than the latest retrieved data
from the same source, then the redundant data SHALL be retrieved and from the same source, then the redundant data <bcp14>SHALL</bcp14> be retriev
appended to the receive buffer. The process SHALL be continued in ed and
appended to the receive buffer. The process <bcp14>SHALL</bcp14> be continued
in
the same way for all younger generations of redundant data. After that, the same way for all younger generations of redundant data. After that,
the timestamp of the packet SHALL be compared with the timestamp of the timestamp of the packet <bcp14>SHALL</bcp14> be compared with the timesta mp of
the latest retrieved data from the same source and if it is later, the latest retrieved data from the same source and if it is later,
then the primary data SHALL be retrieved from the packet and appended then the primary data <bcp14>SHALL</bcp14> be retrieved from the packet and a ppended
to the receive buffer for the source. to the receive buffer for the source.
</t> </t>
</section> </section>
<section numbered="true" toc="default"><name>Delete 'BOM'</name> <section numbered="true" toc="default"><name>Delete BOM</name>
<t> <t>
Unicode character 'BOM' is used as a start indication and sometimes used as a fi ller or keep alive by transmission implementations. These SHALL be deleted after extraction from received packets. The Unicode BOM character is used as a start indication and is sometimes used as a filler or keep-alive by transmission implementations. Any BOM characters <bcp 14>SHALL</bcp14> be deleted after extraction from received packets.
</t> </t>
</section> </section>
</section> </section>
<section anchor="perf" numbered="true" toc="default"> <section anchor="perf" numbered="true" toc="default">
<name>Performance considerations</name> <name>Performance Considerations</name>
<t> <t>
This solution has good performance with low text delays, as long as the me an number of characters per second sent during any 10-second interval from a num ber of simultaneously sending participants to a receiving participant, does not reach the "cps" value. At higher numbers of sent characters per second, a jerkin ess is visible in the presentation of text. The solution is therefore suitable f or emergency service use, relay service use, and small or well-managed larger mu ltimedia conferences. Only in large unmanaged conferences with a high number of participants there may on very rare occasions appear situations when many partic ipants happen to send text simultaneously. In such circumstances, the result may be unpleasantly jerky presentation of text from each sending participant. It sh ould be noted that it is only the number of users sending text within the same m oment that causes jerkiness, not the total number of users with RTT capability. This solution has good performance with low text delays, as long as the me an number of characters per second sent during any 10-second interval from a num ber of simultaneously sending participants to a receiving participant does not r each the "cps" value. At higher numbers of sent characters per second, a jerkine ss is visible in the presentation of text. The solution is therefore suitable fo r emergency service use, relay service use, and small or well-managed larger mul timedia conferences. In large unmanaged conferences with a high number of parti cipants only, on very rare occasions, situations might arise where many particip ants happen to send text simultaneously. In such circumstances, the result may b e unpleasantly jerky presentation of text from each sending participant. It shou ld be noted that it is only the number of users sending text within the same mom ent that causes jerkiness, not the total number of users with real-time text cap ability.
</t> </t>
</section> </section>
<section anchor="security2" numbered="true" toc="default"> <section anchor="security2" numbered="true" toc="default">
<name>Security for session control and media</name> <name>Security for Session Control and Media</name>
<t> <t>
Security mechanisms to provide confidentiality and Security mechanisms to provide confidentiality,
integrity protection and peer authentication SHOULD be applied when possible reg integrity protection, and peer authentication <bcp14>SHOULD</bcp14> be applied w
arding the capabilities of the participating devices by use of SIP over TLS by d hen possible regarding the capabilities of the participating devices by using th
efault according to <xref target="RFC5630" format="default"/> section 3.1.3 on t e Session Initiation Protocol (SIP) over TLS by default according to <xref targe
he session control level and by default using DTLS-SRTP <xref target="RFC5764" f t="RFC5630" sectionFormat="of" section="3.1.3"/>
ormat="default"/> on the media level. In applications where legacy endpoints wit on the session control level and by default using DTLS-SRTP <xref target="RFC576
hout security are allowed, a negotiation SHOULD be performed to decide if encryp 4" format="default"/> at the media level. In applications where legacy endpoints
tion on the media level will be applied. If no other security solution is mandat without security are allowed, a negotiation <bcp14>SHOULD</bcp14> be performed
ed for the application, then OSRTP <xref target="RFC8643" format="default"/> is to decide if encryption at the media level will be applied. If no other security
a suitable method to be applied to negotiate SRTP media security with DTLS. Most solution is mandated for the application, then the Opportunistic Secure Real-ti
SDP examples below are for simplicity expressed without the security additions. me Transport Protocol (OSRTP) <xref target="RFC8643" format="default"/> is a sui
The principles (but not all details) for applying DTLS-SRTP <xref target="RFC57 table method to be applied to negotiate SRTP media security with DTLS. For simpl
64" format="default"/> security are shown in a couple of the following examples. icity, most SDP examples below are expressed without the security additions. The
principles (but not all details) for applying DTLS-SRTP security <xref target="
RFC5764" format="default"/> are shown in a couple of the following examples.
</t> </t>
<t> <t>
Further general security considerations are covered in <xref target="Security" />. Further general security considerations are covered in <xref target="Security" />.
</t> </t>
<t> <t>
End-to-end encryption would require further work and could be based on WebRTC as specified in <xref target="alternatives"/> or on double encryption as specifi ed in <xref target="RFC8723" format="default"/>. End-to-end encryption would require further work and could be based on WebRTC as specified in <xref target="alternatives"/> or on double encryption as specifi ed in <xref target="RFC8723" format="default"/>.
</t> </t>
</section> </section>
<section anchor="SDP-examples2" numbered="true" toc="default"> <section anchor="SDP-examples2" numbered="true" toc="default">
<name>SDP offer/answer examples</name> <name>SDP Offer/Answer Examples</name>
<t> <t>
This section shows some examples of SDP for session negotiation of the rea l-time text media in SIP sessions. Audio is usually provided in the same session , and sometimes also video. The examples only show the part of importance for th e real-time text media. The examples relate to the single RTP stream mixing for multiparty-aware endpoints and for multiparty-unaware endpoints. This section shows some examples of SDP for session negotiation of the rea l-time text media in SIP sessions. Audio is usually provided in the same session , and sometimes also video. The examples only show the part of importance for th e real-time text media. The examples relate to the single RTP stream mixing for multiparty-aware endpoints and for multiparty-unaware endpoints.
</t> </t>
<t> <aside><t>
Note: Multiparty RTT MAY also be provided through other methods, e.g., by a Sele Note: Multiparty real-time text <bcp14>MAY</bcp14> also be provided through othe
ctive Forwarding Middlebox (SFM). In that case, the r methods, e.g., by a Selective Forwarding Middlebox (SFM). In that case, the
SDP of the offer will include something specific for that method, e.g., an SD SDP of the offer will include something specific for that method, e.g., an SD
P attribute or another media format. An answer selecting the use of that method P attribute or another media format. An answer selecting the use of that method
would accept it by a corresponding acknowledgement included in the SDP. The offe would accept it via a corresponding acknowledgement included in the SDP. The off
r may contain also the "rtt-mixer" SDP media attribute for the main RTT media wh er may also contain the "rtt-mixer" SDP media attribute for the main real-time t
en the offeror has capability for both multiparty methods, while an answer, sele ext media when the offerer has this capability for both multiparty methods, whil
cting to use SFM will not include the "rtt-mixer" SDP media attribute. e an answer, choosing to use SFM, will not include the "rtt-mixer" SDP media att
</t> ribute.
</t></aside>
<artwork name="" type="" align="left" alt=""><![CDATA[ <t>Offer example for the "text/red" format, multiparty support, and capability f
or 90 characters per second:</t>
Offer example for "text/red" format and multiparty support: <sourcecode name="" type="sdp"><![CDATA[
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=fmtp:98 cps=90
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
]]></sourcecode>
m=text 11000 RTP/AVP 100 98 <t>Answer example from a multiparty-aware device:</t>
a=rtpmap:98 t140/1000 <sourcecode name="" type="sdp"><![CDATA[
a=rtpmap:100 red/1000 m=text 14000 RTP/AVP 100 98
a=fmtp:100 98/98/98 a=rtpmap:98 t140/1000
a=rtt-mixer a=fmtp:98 cps=90
]]></artwork> a=rtpmap:100 red/1000
<artwork name="" type="" align="left" alt=""><![CDATA[ a=fmtp:100 98/98/98
Answer example from a multiparty-aware device a=rtt-mixer
m=text 14000 RTP/AVP 100 98 ]]></sourcecode>
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
]]></artwork>
<artwork name="" type="" align="left" alt=""><![CDATA[
Offer example for "text/red" format including multiparty <t>Offer example for the "text/red" format, including multiparty and security:</
and security: t>
a=fingerprint: (fingerprint1) <sourcecode name="" type="sdp"><![CDATA[
m=text 11000 RTP/AVP 100 98 a=fingerprint: (fingerprint1)
a=rtpmap:98 t140/1000 m=text 11000 RTP/AVP 100 98
a=rtpmap:100 red/1000 a=rtpmap:98 t140/1000
a=fmtp:100 98/98/98 a=rtpmap:100 red/1000
a=rtt-mixer a=fmtp:100 98/98/98
]]></artwork> a=rtt-mixer
]]></sourcecode>
<t>The "fingerprint" is sufficient to offer DTLS-SRTP, with the media line st ill indicating RTP/AVP. <t>The "fingerprint" is sufficient to offer DTLS-SRTP, with the media line st ill indicating RTP/AVP.
</t> </t>
<t> <aside><t>
Note: For brevity, the entire value of the SDP fingerprint attribute Note: For brevity, the entire value of the SDP "fingerprint" attribute
is not shown in this and the following example. is not shown in this and the following example.
</t> </t></aside>
<artwork name="" type="" align="left" alt=""><![CDATA[ <t>Answer example from a multiparty-aware device with security:</t>
Answer example from a multiparty-aware device with security <sourcecode name="" type="sdp"><![CDATA[
a=fingerprint: (fingerprint2) a=fingerprint: (fingerprint2)
m=text 16000 RTP/AVP 100 98 m=text 16000 RTP/AVP 100 98
a=rtpmap:98 t140/1000 a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000 a=rtpmap:100 red/1000
a=fmtp:100 98/98/98 a=fmtp:100 98/98/98
a=rtt-mixer a=rtt-mixer
]]></artwork> ]]></sourcecode>
<t> <t>
With the "fingerprint" the device acknowledges use of SRTP/DTLS. With the "fingerprint", the device acknowledges the use of DTLS-SRTP.
</t> </t>
<artwork name="" type="" align="left" alt=""><![CDATA[
Answer example from a multiparty-unaware device that also <t>Answer example from a multiparty-unaware device that also
does not support security: does not support security:</t>
<sourcecode name="" type="sdp"><![CDATA[
m=text 12000 RTP/AVP 100 98 m=text 12000 RTP/AVP 100 98
a=rtpmap:98 t140/1000 a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000 a=rtpmap:100 red/1000
a=fmtp:100 98/98/98 a=fmtp:100 98/98/98
]]></artwork> ]]></sourcecode>
</section> </section>
<section anchor="pseq" numbered="true" toc="default"> <section anchor="pseq" numbered="true" toc="default">
<name>Packet sequence example from interleaved transmission</name> <name>Packet Sequence Example from Interleaved Transmission</name>
<t> <t>
This example shows a symbolic flow of packets from a mixer including loss and This example shows a symbolic flow of packets from a mixer, including loss and
recovery. The sequence includes interleaved transmission of text from two RTT so recovery. The sequence includes interleaved transmission of text from two real-
urces A and B. time text sources: A and B.
P indicates primary data. R1 is first redundant generation data and R2 is the P indicates primary data. &nbsp;R1 is the first redundant generation of data,
second redundant generation data. A1, B1, A2 etc. are text chunks (T140blocks) r and R2 is the second redundant generation of data. A1, B1, A2, etc. are text chu
eceived from the respective sources and sent on to the receiver by the mixer. X nks (T140blocks) received from the respective sources and sent on to the receive
indicates a dropped packet between the mixer and a receiver. The session is assu r by the mixer. &nbsp;X indicates a dropped packet between the mixer and a recei
med to use original and two redundant generations of RTT. ver. The session is assumed to use the original and two redundant generations of
real-time text.
</t> </t>
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
|-----------------------| |-----------------------|
|Seq no 101, Time=20400 | |Seq no 101, Time=20400 |
|CC=1 | |CC=1 |
|CSRC list A | |CSRC list A |
|R2: A1, Offset=600 | |R2: A1, Offset=600 |
|R1: A2, Offset=300 | |R1: A2, Offset=300 |
|P: A3 | |P: A3 |
|-----------------------| |-----------------------|
]]></artwork> ]]></artwork>
<t> <t>
Assuming that earlier packets (with text A1 and A2) were received in sequence, text A3 is received from packet 101 and assigned to reception buffer A. The mix er is now assumed to have received initial text from source B 100 ms after packe t 101 and will send that text. Transmission of A2 and A3 as redundancy is planne d for 330 ms after packet 101 if no new text from A is ready to be sent before t hat. Assuming that earlier packets (with text A1 and A2) were received in sequence, text A3 is received from packet 101 and assigned to reception buffer A. The mix er is now assumed to have received initial text from source B 100 ms after packe t 101 and will send that text. Transmission of A2 and A3 as redundancy is planne d for 330 ms after packet 101 if no new text from A is ready to be sent before t hat.
</t> </t>
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
|-----------------------| |-----------------------|
|Seq no 102, Time=20500 | |Seq no 102, Time=20500 |
|CC=1 | |CC=1 |
|CSRC list B | |CSRC list B |
|R2 Empty, Offset=600 | |R2 Empty, Offset=600 |
|R1: Empty, Offset=300 | |R1: Empty, Offset=300 |
|P: B1 | |P: B1 |
|-----------------------| |-----------------------|
Packet 102 is received.
B1 is retrieved from this packet. Redundant transmission of
B1 is planned 330 ms after packet 102.
]]></artwork> ]]></artwork>
<t indent="3">Packet 102 is received.</t>
<t indent="3">B1 is retrieved from this packet. Redundant transmission of
B1 is planned 330 ms after packet 102.</t>
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
X------------------------| X------------------------|
X Seq no 103, Timer=20730| X Seq no 103, Timer=20730|
X CC=1 | X CC=1 |
X CSRC list A | X CSRC list A |
X R2: A2, Offset=630 | X R2: A2, Offset=630 |
X R1: A3, Offset=330 | X R1: A3, Offset=330 |
X P: Empty | X P: Empty |
X------------------------| X------------------------|
Packet 103 is assumed to be lost due to network problems.
It contains redundancy for A. Sending A3 as second level
redundancy is planned for 330 ms after packet 103.
]]></artwork> ]]></artwork>
<t indent="3">Packet 103 is assumed to be lost due to network problems.</t>
<t indent="3">It contains redundancy for A. Sending A3 as second-level
redundancy is planned for 330 ms after packet 103.</t>
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
X------------------------| X------------------------|
X Seq no 104, Timer=20800| X Seq no 104, Timer=20800|
X CC=1 | X CC=1 |
X CSRC list B | X CSRC list B |
X R2: Empty, Offset=600 | X R2: Empty, Offset=600 |
X R1: B1, Offset=300 | X R1: B1, Offset=300 |
X P: B2 | X P: B2 |
X------------------------| X------------------------|
Packet 104 contains text from B, including new B2 and ]]></artwork>
<t indent="3">Packet 104 contains text from B, including new B2 and
redundant B1. It is assumed dropped due to network redundant B1. It is assumed dropped due to network
problems. problems.</t>
The mixer has A3 redundancy to send, but no new text <t indent="3">The mixer has A3 redundancy to send, but no new text
appears from A and therefore the redundancy is sent appears from A, and therefore the redundancy is sent
330 ms after the previous packet with text from A. 330 ms after the previous packet with text from A.</t>
]]></artwork>
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
|------------------------| |------------------------|
| Seq no 105, Timer=21060| | Seq no 105, Timer=21060|
| CC=1 | | CC=1 |
| CSRC list A | | CSRC list A |
| R2: A3, Offset=660 | | R2: A3, Offset=660 |
| R1: Empty, Offset=330 | | R1: Empty, Offset=330 |
| P: Empty | | P: Empty |
|------------------------| |------------------------|
Packet 105 is received. ]]></artwork>
A gap for lost packets 103 and 104 is detected.
<t indent="3">Packet 105 is received.</t>
<t indent="3">A gap for lost packets 103 and 104 is detected.
Assume that no other loss was detected during the last second. Assume that no other loss was detected during the last second.
Then it can be concluded that nothing was totally lost. It can then be concluded that nothing was totally lost.</t>
R2 is checked. Its original time was 21060-660=20400. <t indent="3">R2 is checked. Its original time was 21060-660=20400.
A packet with text from A was received with that A packet with text from A was received with that
timestamp, so nothing needs to be recovered. timestamp, so nothing needs to be recovered.</t>
<t indent="3">B1 and B2 still need to be transmitted as redundancy.
This is planned 330 ms after packet 104. That would be at 21130.</t>
B1 and B2 still need to be transmitted as redundancy.
This is planned 330 ms after packet 104. That
would be at 21130.
]]></artwork>
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
|-----------------------| |-----------------------|
|Seq no 106, Timer=21130| |Seq no 106, Timer=21130|
|CC=1 | |CC=1 |
|CSRC list B | |CSRC list B |
| R2: B1, Offset=630 | | R2: B1, Offset=630 |
| R1: B2, Offset=330 | | R1: B2, Offset=330 |
| P: Empty | | P: Empty |
|-----------------------| |-----------------------|
]]></artwork> ]]></artwork>
<t>
Packet 106 is received. <t indent="3">Packet 106 is received.</t>
</t> <t indent="3">The second-level redundancy in packet 106 is B1 and has a timest
<t> amp offset of 630 ms.
The second level redundancy in packet 106 is B1 and has timestamp offset 630 ms. The timestamp of packet 106 minus 630 is 20500, which is the timestamp of packet
The timestamp of packet 106 minus 630 is 20500 which is the timestamp of packet 102 that was received. So, B1 does not need to be retrieved.
102 that was received. So B1 does not need to be retrieved. The first-level redundancy in packet 106 has an offset of 330. The timestamp of
The first level redundancy in packet 106 has offset 330. The timestamp of packet packet 106 minus 330 is 20800. That is later than the latest received packet wit
106 minus 330 is 20800. That is later than the latest received packet with sour h source B. Therefore, B2 is retrieved and assigned to the input buffer for sour
ce B. Therefore B2 is retrieved and assigned to the input buffer for source B. ce B.
No primary is available in packet 106. No primary is available in packet 106.</t>
</t> <t indent="3">After this sequence, A3, B1, and B2 have been received. In this
<t> case, no text was lost.</t>
After this sequence, A3 and B1 and B2 have been received. In this case no te
xt was lost.
</t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Maximum character rate "cps"</name> <name>Maximum Character Rate "cps" Setting</name>
<t> <t>
The default maximum rate of reception of "text/t140" real-time text is The default maximum rate of reception of "text/t140" real-time text, a
in <xref target="RFC4103" format="default"/> specified to be 30 characters per s s specified in <xref target="RFC4103" format="default"/>, is 30 characters per s
econd. The actual rate is calculated without regard to any redundant text transm econd. The actual rate is calculated without regard to any redundant text transm
ission and is in the multiparty case evaluated for all sources contributing to t ission and is, in the multiparty case, evaluated for all sources contributing to
ransmission to a receiver. The value MAY be modified in the "cps" parameter of t transmission to a receiver. The value <bcp14>MAY</bcp14> be modified in the "cp
he FMTP attribute in the media section for the "text/t140" media. A mixer combin s" parameter of the "fmtp" attribute for the "text/t140" format of the "text" me
ing real-time text from a number of sources may occasionally have a higher combi dia section.
ned flow of text coming from the sources. Endpoints SHOULD therefore specify a s </t>
uitable higher value for the "cps" parameter, corresponding to its real receptio <t>
n capability. A value for "cps" of 90 SHALL be the default for the "text/t140" s A mixer combining real-time text from a number of sources may o
tream in the "text/red" format when multiparty real-time text is negotiated. See ccasionally have a higher combined flow of text coming from the sources. Endpoin
<xref target="RFC4103" format="default"/> for the format and use of the "cps" p ts <bcp14>SHOULD</bcp14> therefore include a suitable higher value for the "cps"
arameter. The same rules apply for the multiparty case except for the default va parameter, corresponding to its real reception capability. The default "cps" va
lue. lue 30 can be assumed to be sufficient for small meetings and well-managed large
</t> r conferences with users only making manual text entry. A "cps" value of 90 can
be assumed to be sufficient even for large unmanaged conferences and for cases w
hen speech-to-text technologies are used for text entry. This is also a reachabl
e performance for receivers in modern technologies, and 90 is therefore the <bcp
14>RECOMMENDED</bcp14> "cps" value. See <xref target="RFC4103" format="default"/
> for the format and use of the "cps" parameter. The same rules apply for the mu
ltiparty case.
</t>
</section> </section>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Presentation level considerations</name> <name>Presentation-Level Considerations</name>
<t> <t>
"Protocol for multimedia application text conversation" <xref target="T140" form "Protocol for multimedia application text conversation" <xref target="T140" form
at="default"/> provides the presentation level requirements for the <xref target at="default"/> provides the presentation-level requirements for RTP transport
="RFC4103" format="default"/> transport. Functions for erasure and other formatt as described in <xref target="RFC4103" format="default"/>. Functions for erasure
ing functions are specified in <xref target="T140" format="default"/> which has and other formatting functions are specified in <xref target="T140" format="def
the following general statement for the presentation: ault"/>, which has the following general statement for the presentation:
</t> </t>
<t> <!-- DNE; verified -->
"The display of text from the members of the conversation should be arranged so <blockquote>The display of text from the members of the conversation should be a
that the text from rranged so that the text from
each participant is clearly readable, and its source and the relative timing of entered text is visualized each participant is clearly readable, and its source and the relative timing of entered text is visualized
in the display. Mechanisms for looking back in the contents from the current ses sion should be in the display. Mechanisms for looking back in the contents from the current ses sion should be
provided. The text should be displayed as soon as it is received." provided. The text should be displayed as soon as it is received.</blockquote>
</t>
<t> <t>
Strict application of <xref target="T140" format="default"/> is of essence for t he interoperability of real-time text implementations and to fulfill the intenti on that the session participants have the same information conveyed in the text contents of the conversation without necessarily having the exact same layout of the conversation. Strict application of <xref target="T140" format="default"/> is essential for th e interoperability of real-time text implementations and to fulfill the intentio n that the session participants have the same information conveyed in the text c ontents of the conversation without necessarily having the exact same layout of the conversation.
</t> </t>
<t> <t>
<xref target="T140" format="default"/> specifies a set of presentation control codes to include in the stream. Some of them are optional. Implementations MUST ignore optional control codes that they do not support. <xref target="T140" format="default"/> specifies a set of presentation control codes (<xref target="act-tx-ctrl-codes"/>) to include in the stream. Some of the m are optional. Implementations <bcp14>MUST</bcp14> ignore optional control code s that they do not support.
</t> </t>
<t> <t>
There is no strict "message" concept in real-time text. The Unicode Line Separat or character SHALL be used as a separator allowing a part of received text to be grouped in presentation. The characters "CRLF" may be used by other implementat ions as a replacement for Line Separator. The "CRLF" combination SHALL be erased by just one erasing action, the same as the Line Separator. Presentation functi ons are allowed to group text for presentation in smaller groups than the line s eparators imply and present such groups with source indication together with tex t groups from other sources (see the following presentation examples). Erasure h as no specific limit by any delimiter in the text stream. There is no strict "message" concept in real-time text. The Unicode Line Separat or character <bcp14>SHALL</bcp14> be used as a separator allowing a part of rece ived text to be grouped in a presentation. The character combination "CRLF" may be used by other implementations as a replacement for the Line Separator. The "C RLF" combination <bcp14>SHALL</bcp14> be erased by just one erasing action, the same as the Line Separator. Presentation functions are allowed to group text for presentation in smaller groups than the Line Separators imply and present such groups with a source indication together with text groups from other sources (se e the following presentation examples). Erasure has no specific limit by any del imiter in the text stream.
</t> </t>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Presentation by multiparty-aware endpoints</name> <name>Presentation by Multiparty-Aware Endpoints</name>
<t> <t>
A multiparty-aware receiving party, presenting real-time text MUST separate text A multiparty-aware receiving party presenting real-time text <bcp14>MUST</bcp14>
from different sources and present them in separate presentation fields. separate text from different sources and present them in separate presentation
The receiving party MAY separate presentation of parts of text from a source in fields.
readable groups based on other criteria than line separator and merge these grou The receiving party <bcp14>MAY</bcp14> separate the presentation of parts of tex
ps in the presentation area when it benefits the user to most easily find and re t from a source in readable groups based on criteria other than a Line Separator
ad text from the different participants. The criteria MAY e.g., be a received co and merge these groups in the presentation area when it benefits the user to mo
mma, full stop, or other phrase delimiters, or a long pause. st easily find and read text from the different participants. The criteria <bcp1
4>MAY</bcp14>, for example, be a received comma, a full stop, some other type of
phrase delimiter, or a long pause.
</t> </t>
<t> <t>
When text is received from multiple original sources, the presentation SHALL pro vide a view where text is added in multiple presentation fields. When text is received from multiple original sources, the presentation <bcp14>SH ALL</bcp14> provide a view where text is added in multiple presentation fields.
</t> </t>
<t> <t>
If the presentation presents text from different sources in one common area, the If the presentation presents text from different sources in one common area, the
presenting endpoint SHOULD insert text from the local user ended at suitable po presenting endpoint <bcp14>SHOULD</bcp14> insert text from the local user, wher
ints merged with received text to indicate the relative timing for when the text e the text ends at suitable points and is merged properly with received text to
groups were completed. In this presentation mode, the receiving endpoint SHALL indicate the relative timing for when the text groups were completed.
present the source of the different groups of text. This presentation style is c In this presentation mode, the receiving endpoint <bcp14>SHALL</bcp14> present
alled the "chat" style here and provides a possibility to follow text arriving f the source of the different groups of text. This presentation style is called th
rom multiple parties and the approximate relative time that text is received rel e "chat" style here and provides the possibility of following text arriving from
ated to text from the local user. multiple parties and the approximate relative time that text is received as rel
ated to text from the local user.
</t> </t>
<t>A view of a three-party RTT call in chat style is shown in this examp <t>A view of a three-party real-time text call in chat style is shown in
le .</t> this example.</t>
<figure anchor="fig1" title="Example of a Three-Party Real-Time Text Ca
ll Presented in Chat Style Seen at Participant Alice's Endpoint">
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
_________________________________________________ _________________________________________________
| |^| | |^|
|[Alice] Hi, Alice here. |-| |[Alice] Hi, Alice here. |-|
| | | | | |
|[Bob] Bob as well. | | |[Bob] Bob as well. | |
| | | | | |
|[Eve] Hi, this is Eve, calling from Paris. | | |[Eve] Hi, this is Eve, calling from Paris. | |
| I thought you should be here. | | | I thought you should be here. | |
| | | | | |
|[Alice] I am coming on Thursday, my | | |[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| | | performance is not until Friday morning.| |
| | | | | |
|[Bob] And I on Wednesday evening. | | |[Bob] And I on Wednesday evening. | |
| | | | | |
|[Alice] Can we meet on Thursday evening? | | |[Alice] Can we meet on Thursday evening? | |
| | | | | |
|[Eve] Yes, definitely. How about 7pm. | | |[Eve] Yes, definitely. How about 7pm. | |
| at the entrance of the restaurant | | | at the entrance of the restaurant | |
| Le Lion Blanc? | | | Le Lion Blanc? | |
|[Eve] we can have dinner and then take a walk |-| |[Eve] we can have dinner and then take a walk |-|
|______________________________________________|v| |______________________________________________|v|
| <Eve-typing> But I need to be back to |^| | <Eve-typing> But I need to be back to |^|
| the hotel by 11 because I need |-| | the hotel by 11 because I need |-|
| | | | | |
| <Bob-typing> I wou |-| | <Bob-typing> I wou |-|
|______________________________________________|v| |______________________________________________|v|
| of course, I underst | | of course, I underst |
|________________________________________________| |________________________________________________|
]]></artwork> ]]></artwork>
</figure>
<t> Figure 3: Example of a three-party RTT call presented in chat style <t>Presentation styles other than the chat style <bcp14>MAY</bcp14> be a
seen at participant 'Alice's endpoint. rranged.
</t>
<t>Other presentation styles than the chat style MAY be arranged.
</t> </t>
<t>This figure shows how a coordinated column view <t><xref target="fig2"/> shows how a coordinated column view
MAY be presented.</t> <bcp14>MAY</bcp14> be presented.</t>
<artwork name="" type="" align="left" alt=""><![CDATA[__________________ <figure anchor="fig2" title="An Example of a Coordinated Column View of
___________________________________________________ a Three&nbhy;Party Session with Entries Ordered Vertically in Approximate Time O
rder">
<artwork name="" type="" align="left" alt=""><![CDATA[
_____________________________________________________________________
| Bob | Eve | Alice | | Bob | Eve | Alice |
|____________________|______________________|_______________________| |____________________|______________________|_______________________|
| | |I will arrive by TGV. | | | |I will arrive by TGV. |
|My flight is to Orly| |Convenient to the main | |My flight is to Orly| |Convenient to the main |
| |Hi all, can we plan |station. | | |Hi all, can we plan |station. |
| |for the seminar? | | | |for the seminar? | |
|Eve, will you do | | | |Eve, will you do | | |
|your presentation on| | | |your presentation on| | |
|Friday? |Yes, Friday at 10. | | |Friday? |Yes, Friday at 10. | |
|Fine, wo | |We need to meet befo | |Fine, wo | |We need to meet befo |
|___________________________________________________________________| |___________________________________________________________________|
]]></artwork> ]]></artwork>
<t>Figure 4: An example of a coordinated column-view of a three-party </figure>
session with entries ordered vertically in approximate time-order.</t>
</section> </section>
<section anchor="UnawareFormat" numbered="true" toc="default"> <section anchor="UnawareFormat" numbered="true" toc="default">
<name>Multiparty mixing for multiparty-unaware endpoints</name> <name>Multiparty Mixing for Multiparty-Unaware Endpoints</name>
<t> <t>
When the mixer has indicated RTT multiparty capability in an SDP negotiatio n, but the multiparty capability negotiation fails with an endpoint, then the ag reed "text/red" or "text/t140" format SHALL be used and the mixer SHOULD compose a best-effort presentation of multiparty real-time text in one stream intended to be presented by an endpoint with no multiparty awareness, when that is desire d in the actual implementation. The following specifies a procedure which MAY be applied in that situation. When the mixer has indicated multiparty real-time text capability in an SDP negotiation but the multiparty capability negotiation fails with an endpoint, th e agreed-upon "text/red" or "text/t140" format <bcp14>SHALL</bcp14> be used and the mixer <bcp14>SHOULD</bcp14> compose a best-effort presentation of multiparty real-time text in one stream intended to be presented by an endpoint with no mu ltiparty awareness, when that is desired in the actual implementation. The follo wing specifies a procedure that <bcp14>MAY</bcp14> be applied in that situation.
</t> </t>
<t> <t>
This presentation format has functional limitations and SHOULD be used on ly to enable participation in multiparty calls by legacy deployed endpoints impl ementing only RFC 4103 without any multiparty extensions specified in this docum ent. This presentation format has functional limitations and <bcp14>SHOULD</b cp14> be used only to enable participation in multiparty calls by legacy deploye d endpoints implementing only RFC 4103 without any multiparty extensions specifi ed in this document.
</t> </t>
<t> <t>
The principles and procedures below do not specify any new protocol eleme nts. They are instead composed of information from <xref target="T140" format="d efault"/> and an ambition to provide a best-effort presentation on an endpoint w hich has functions originally intended only for two-party calls. The principles and procedures below do not specify any new protocol elem ents. They are instead composed of information provided in <xref target="T140" f ormat="default"/> and an ambition to provide a best-effort presentation on an en dpoint that has functions originally intended only for two-party calls.
</t> </t>
<t> <t>
The mixer mixing for multiparty-unaware endpoints SHALL compose a simulated, lim The mixer performing the mixing for multiparty-unaware endpoints <bcp14>SHALL</b
ited multiparty RTT view suitable for presentation in one presentation area. The cp14> compose a simulated, limited multiparty real-time text view suitable for p
mixer SHALL group text in suitable groups and prepare for presentation of them resentation in one presentation area.
by inserting a line separator between them if the transmitted text did not alrea The mixer <bcp14>SHALL</bcp14> group text in suitable groups and prepare them f
dy end with a new line (line separator or CRLF). A presentable label SHALL be co or presentation by inserting a Line Separator between them if the transmitted te
mposed and sent for the source initially in the session and after each source sw xt did not already end with a new line (Line Separator or CRLF). A presentable l
itch. With this procedure the time for switching from transmission of text from abel <bcp14>SHALL</bcp14> be composed and sent for the source initially in the s
one source to transmission of text from another source depends on the actions of ession and after each source switch. With this procedure, the time for switching
the users. In order to expedite source switching, a user can, for example, end from transmission of text from one source to transmission of text from another
its turn with a new line. source depends on the actions of the users. In order to expedite source switchin
g, a user can, for example, end its turn with a new line.
</t> </t>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Actions by the mixer at reception from the call participants</na me> <name>Actions by the Mixer at Reception from the Call Participants</na me>
<t> <t>
When text is received by the mixer from the different participants, the mixer SH When text is received by the mixer from the different participants, the mixer <b
ALL recover text from redundancy if any packets are lost. The mark for lost text cp14>SHALL</bcp14> recover text from redundancy if any packets are lost. The mar
<xref target="T140ad1" format="default"/> SHALL be inserted in the stream if un ker for lost text <xref target="T140ad1" format="default"/> <bcp14>SHALL</bcp14>
recoverable loss appears. Any Unicode "BOM" characters, possibly used for keep-a be inserted in the stream if unrecoverable loss appears. Any Unicode BOM charac
live, SHALL be deleted. ters, possibly used for keep-alives, <bcp14>SHALL</bcp14> be deleted.
The time of creation of text (retrieved from the RTP timestamp) SHALL be stored The time of creation of text (retrieved from the RTP timestamp) <bcp14>SHALL</bc
together with the received text from each source in queues for transmission to t p14> be stored together with the received text from each source in queues for tr
he recipients in order to be able to evaluate text loss. ansmission to the recipients in order to be able to evaluate text loss.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Actions by the mixer for transmission to the recipients</name> <name>Actions by the Mixer for Transmission to the Recipients</name>
<t> <t>
The following procedure SHALL be applied for each multiparty-unaware recipient o f multiparty text from the mixer. The following procedure <bcp14>SHALL</bcp14> be applied for each multiparty-unaw are recipient of multiparty text from the mixer.
</t> </t>
<t> <t>
The text for transmission SHALL be formatted by the mixer for each receiving use r for presentation in one single presentation area. Text received from a partici pant SHOULD NOT be included in transmission to that participant because it is us ually presented locally at transmission time. When there is text available for t ransmission from the mixer to a receiving party from more than one participant, the mixer SHALL switch between transmission of text from the different sources a t suitable points in the transmitted stream. The text for transmission <bcp14>SHALL</bcp14> be formatted by the mixer for eac h receiving user for presentation in one single presentation area. Text received from a participant <bcp14>SHOULD NOT</bcp14> be included in transmissions to th at participant, because it is usually presented locally at transmission time. Wh en there is text available for transmission from the mixer to a receiving party from more than one participant, the mixer <bcp14>SHALL</bcp14> switch between tr ansmission of text from the different sources at suitable points in the transmit ted stream.
</t> </t>
<t> <t>
When switching source, the mixer SHALL insert a line separator if the already tr ansmitted text did not end with a new line (line separator or CRLF). A label SHA LL be composed of information in the CNAME and NAME fields in RTCP reports from the participant to have its text transmitted, or from other session information for that user. The label SHALL be delimited by suitable characters (e.g., '[ ]') and transmitted. The CSRC SHALL indicate the selected source. Then text from th at selected participant SHALL be transmitted until a new suitable point for swit ching source is reached. When switching the source, the mixer <bcp14>SHALL</bcp14> insert a Line Separato r if the already-transmitted text did not end with a new line (Line Separator or CRLF). A label <bcp14>SHALL</bcp14> be composed of information in the CNAME and NAME fields in RTCP reports from the participant to have its text transmitted, or from other session information for that user. The label <bcp14>SHALL</bcp14> be delimited by suitable characters (e.g., "[&nbsp;]") and transmitted. The CSRC <bcp14>SHALL</bcp14> indicate the selected source. Then, text from that selecte d participant <bcp14>SHALL</bcp14> be transmitted until a new suitable point for switching the source is reached.
</t> </t>
<t> <t>
Information available to the mixer for composing the label may contain sensiti ve personal information that SHOULD NOT be revealed in sessions not securely aut henticated and confidentiality protected. Privacy considerations regarding how m uch personal information is included in the label SHOULD therefore be taken when composing the label. Information available to the mixer for composing the label may contain sensiti ve personal information that <bcp14>SHOULD NOT</bcp14> be revealed in sessions n ot securely authenticated and confidentiality protected. Privacy considerations regarding how much personal information is included in the label <bcp14>SHOULD</ bcp14> therefore be taken when composing the label.
</t> </t>
<t> <t>
Seeking a suitable point for switching source SHALL be done when there is older text waiting for transmission from any party than the age of the last transmitte d text. Seeking a suitable point for switching the source <bcp14>SHALL</bcp14> be done w hen there is older text waiting for transmission from any party than the age of the last transmitted text.
Suitable points for switching are:</t> Suitable points for switching are:</t>
<ul> <ul>
<li>A completed phrase ended by comma</li> <li>A completed phrase ending with a comma.</li>
<li>A completed sentence</li> <li>A completed sentence.</li>
<li>A new line (line separator or CRLF)</li> <li>A new line (Line Separator or CRLF).</li>
<li>A long pause (e.g., &gt; 10 seconds) in received text from the c <li>A long pause (e.g., &gt; 10 seconds) in received text from the c
urrently transmitted source</li> urrently transmitted source.</li>
<li>If text from one participant has been transmitted with text from <li>If text from one participant has been transmitted with text from
other sources waiting for transmission for a long time (e.g., &gt; 1 minute) an other sources waiting for transmission for a long time (e.g., &gt; 1 minute) an
d none of the other suitable points for switching has occurred, a source switch d none of the other suitable points for switching has occurred, a source switch
MAY be forced by the mixer at the next word delimiter, and also even if a word <bcp14>MAY</bcp14> be forced by the mixer at the next word delimiter, and also e
delimiter does not occur within a time (e.g., 15 seconds) after the scan for a w ven if a word delimiter does not occur within some period of time (e.g., 15 seco
ord delimiter started.</li> nds) after the scan for a word delimiter started.</li>
</ul> </ul>
<t> <t>
When switching source, the source which has the oldest text in queue SHALL be se When switching the source, the source that has the oldest text in queue <bcp14>S
lected to be transmitted. HALL</bcp14> be selected to be transmitted.
A character display count SHALL be maintained for the currently transmitted sour A character display count <bcp14>SHALL</bcp14> be maintained for the currently t
ce, starting at zero after the label is transmitted for the currently transmitte ransmitted source, starting at zero after the label is transmitted for the curre
d source. ntly transmitted source.
</t> </t>
<t> <t>
The status SHALL be maintained for the latest control code for Select Graphic Re The status <bcp14>SHALL</bcp14> be maintained for the latest control code for Se
ndition (SGR) from each source. lect Graphic Rendition (SGR) from each source.
If there is an SGR code stored as the status for the current source before the s If there is an SGR code stored as the status for the current source before the s
ource switch is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B 0 ource switch is done, a reset of SGR <bcp14>SHALL</bcp14> be sent by the sequenc
000 006D] after the new line and before the new label during a source switch. Se e SGR 0 [U+009B U+0000 U+006D] after the new line and before the new label durin
e SGR below for an explanation. This transmission does not influence the display g a source switch. See <xref target="act-tx-ctrl-codes"/> for an explanation.
count. This transmission does not influence the display count.
</t> </t>
<t> <t>
If there is an SGR code stored for the new source after the source switch, that SGR code SHALL be transmitted to the recipient before the label. This transmissi on does not influence the display count. If there is an SGR code stored for the new source after the source switch, that SGR code <bcp14>SHALL</bcp14> be transmitted to the recipient before the label. This transmission does not influence the display count.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Actions on transmission of text</name> <name>Actions on Transmission of Text</name>
<t> <t>
Text from a source sent to the recipient SHALL increase the display count by one per transmitted character. Text from a source sent to the recipient <bcp14>SHALL</bcp14> increase the displ ay count by one per transmitted character.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section anchor="act-tx-ctrl-codes" numbered="true" toc="default">
<name>Actions on transmission of control codes</name> <name>Actions on Transmission of Control Codes</name>
<t> <t>
The following control codes specified by T.140 require specific actions. They SH ALL cause specific considerations in the mixer. Note that the codes presented he re are expressed in UCS-16, while transmission is made in the UTF-8 encoding of these codes. The following control codes, as specified by T.140 <xref target="T140" format="d efault"/>, require specific actions. They <bcp14>SHALL</bcp14> cause specific co nsiderations in the mixer. Note that the codes presented here are expressed in U TF-16, while transmission is made in the UTF-8 encoding of these codes.
</t> </t>
<dl> <dl>
<dt>BEL 0007 Bell</dt> <dt>BEL (U+0007):</dt>
<dd> Alert in session. Provides for alerting during an active sessio <dd>Bell. Alert in session. Provides for alerting during an active s
n. The display count SHALL NOT be altered.</dd> ession. The display count <bcp14>SHALL NOT</bcp14> be altered.</dd>
<dt>NEW LINE 2028</dt> <dt>NEW LINE (U+2028):</dt>
<dd> Line separator. Check and perform a source switch if appropriat <dd>Line Separator. Check and perform a source switch if appropriate
e. Increase the display count by 1.</dd> . Increase the display count by 1.</dd>
<dt>CR LF 000D 000A</dt> <dt>CR LF (U+000D U+000A):</dt>
<dd> A supported but not preferred way of requesting a new line. Che <dd>A supported, but not preferred, way of requesting a new line. Ch
ck and perform a source switch if appropriate. Increase the display count by 1.< eck and perform a source switch if appropriate. Increase the display count by 1.
/dd> </dd>
<dt>INT ESC 0061</dt> <dt>INT (ESC U+0061):</dt>
<dd> Interrupt (used to initiate the mode negotiation procedure). Th <dd>Interrupt (used to initiate the mode negotiation procedure). The
e display count SHALL NOT be altered.</dd> display count <bcp14>SHALL NOT</bcp14> be altered.</dd>
<dt>SGR 009B Ps 006D </dt> <dt>SGR (U+009B Ps U+006D):</dt>
<dd>Select graphic rendition. Ps is the rendition parameters specif <dd>Select Graphic Rendition. Ps represents the rendition parameter
ied in ISO 6429. The display count SHALL NOT be altered. The SGR code SHOULD be s specified in <xref target="ISO6429" format="default"/>. (For freely available
stored for the current source.</dd> equivalent information, please see <xref target="ECMA-48"/>.)
<dt>SOS 0098</dt> The display count <bcp14>SHALL NOT</bcp14> be altered. The SGR code <bcp14>SHOU
<dd> Start of string, used as a general protocol element introducer, LD</bcp14> be stored for the current source.</dd>
followed by a maximum 256-byte string and the ST. The display count SHALL NOT b <dt>SOS (U+0098):</dt>
e altered.</dd> <dd> Start of String. Used as a general protocol element introducer,
<dt>ST 009C</dt> followed by a maximum 256-byte string and the ST. The display count <bcp14>SHAL
<dd> String terminator, end of SOS string. The display count SHALL N L NOT</bcp14> be altered.</dd>
OT be altered.</dd> <dt>ST (U+009C):</dt>
<dt>ESC 001B</dt> <dd> String Terminator. End of SOS string. The display count <bcp14>
<dd> Escape - used in control strings. The display count SHALL NOT b SHALL NOT</bcp14> be altered.</dd>
e altered for the complete escape code.</dd> <dt>ESC (U+001B):</dt>
<dt>Byte order mark "BOM" (U+FEFF)</dt> <dd> Escape. Used in control strings. The display count <bcp14>SHAL
<dd>"Zero width, no break space", used for synchronization and keep- L NOT</bcp14> be altered for the complete escape code.</dd>
alive. It SHALL be deleted from incoming streams. It SHALL also be sent first af <dt>Byte order mark (BOM) (U+FEFF):</dt>
ter session establishment to the recipient. The display count SHALL NOT be alter <dd>"Zero width no-break space". Used for synchronization and keep-a
ed.</dd> live. It <bcp14>SHALL</bcp14> be deleted from incoming streams. It <bcp14>SHALL<
<dt>Missing text mark (U+FFFD)</dt> /bcp14> also be sent first after session establishment to the recipient. The dis
<dd> "Replacement character", represented as a question mark in a rh play count <bcp14>SHALL NOT</bcp14> be altered.</dd>
ombus, or if that is not feasible, replaced by an apostrophe '. It marks the pla <dt>Missing text mark (U+FFFD):</dt>
ce in the stream of possible text loss. This mark SHALL be inserted by the recep <dd>"Replacement character". Represented as a question mark in a rho
tion procedure in case of unrecoverable loss of packets. The display count SHALL mbus, or, if that is not feasible, replaced by an apostrophe ('). It marks the p
be increased by one when sent as for any other character.</dd> lace in the stream of possible text loss. This mark <bcp14>SHALL</bcp14> be inse
<dt>SGR</dt> rted by the reception procedure in the case of unrecoverable loss of packets. Th
<dd> If a control code for selecting graphic rendition (SGR) other t e display count <bcp14>SHALL</bcp14> be increased by one when sent as for any ot
han reset of the graphic rendition (SGR 0) is sent to a recipient, that control her character.</dd>
code SHALL also be stored as the status for the source in the storage for SGR st <dt>SGR:</dt>
atus. If a reset graphic rendition (SGR 0) originating from a source is sent, th <dd>If a control code for SGR other than a reset of the graphic rend
en the SGR status storage for that source SHALL be cleared. The display count S ition (SGR 0) is sent to a recipient, that control code <bcp14>SHALL</bcp14> als
HALL NOT be increased.</dd> o be stored as the status for the source in the storage for SGR status. If a res
<dt>BS (U+0008)</dt> et graphic rendition (SGR 0) originating from a source is sent, then the SGR sta
<dd> Back Space, intended to erase the last entered character by a s tus storage for that source <bcp14>SHALL</bcp14> be cleared. The display count
ource. Erasure by backspace cannot always be performed as the erasing party inte <bcp14>SHALL NOT</bcp14> be increased.</dd>
nded. If an erasing action erases all text up to the end of the leading label a <dt>BS (U+0008):</dt>
fter a source switch, then the mixer MUST NOT transmit more backspaces. Instead, <dd>"Back Space". Intended to erase the last entered character by a
it is RECOMMENDED that a letter "X" is inserted in the text stream for each bac source. Erasure by backspace cannot always be performed as the erasing party int
kspace as an indication of the intent to erase more. A new line is usually coded ended. If an erasing action erases all text up to the end of the leading label
by a Line Separator, but the character combination "CRLF" MAY be used instead. after a source switch, then the mixer <bcp14>MUST NOT</bcp14> transmit more back
Erasure of a new line is in both cases done by just one erasing action (Backspa spaces. Instead, it is <bcp14>RECOMMENDED</bcp14> that a letter "X" be inserted
ce). If the display count has a positive value it SHALL be decreased by one when in the text stream for each backspace as an indication of the intent to erase mo
the BS is sent. If the display count is at zero, it SHALL NOT be altered.</dd> re. A new line is usually coded by a Line Separator, but the character combinati
on "CRLF" <bcp14>MAY</bcp14> be used instead. Erasure of a new line is, in both
cases, done by just one erasing action (backspace). If the display count has a
positive value, it <bcp14>SHALL</bcp14> be decreased by one when the BS is sent.
If the display count is at zero, it <bcp14>SHALL NOT</bcp14> be altered.</dd>
</dl> </dl>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Packet transmission</name> <name>Packet Transmission</name>
<t> <t>
A mixer transmitting to a multiparty-unaware terminal SHALL send primary dat a only from one source per packet. The SSRC SHALL be the SSRC of the mixer. The CSRC list SHALL contain one member and be the SSRC of the source of the primary data. A mixer transmitting to a multiparty-unaware endpoint <bcp14>SHALL</bcp14> s end primary data only from one source per packet. The SSRC <bcp14>SHALL</bcp14> be the SSRC of the mixer. The CSRC list <bcp14>MAY</bcp14> contain one member an d be the SSRC of the source of the primary data.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Functional limitations</name> <name>Functional Limitations</name>
<t> <t>
When a multiparty-unaware endpoint presents a conversation in one display ar ea in a chat style, it inserts source indications for remote text and local user text as they are merged in completed text groups. When an endpoint using this l ayout receives and presents text mixed for multiparty-unaware endpoints, there w ill be two levels of source indicators for the received text; one generated by t he mixer and inserted in a label after each source switch, and another generated by the receiving endpoint and inserted after each switch between local and remo te source in the presentation area. This will waste display space and look incon sistent to the reader. When a multiparty-unaware endpoint presents a conversation in one display ar ea in a chat style, it inserts source indications for remote text and local user text as they are merged in completed text groups. When an endpoint using this l ayout receives and presents text mixed for multiparty-unaware endpoints, there w ill be two levels of source indicators for the received text: one generated by t he mixer and inserted in a label after each source switch, and another generated by the receiving endpoint and inserted after each switch between the local sour ce and the remote source in the presentation area. This will waste display space and look inconsistent to the reader.
</t> </t>
<t> <t>
New text can be presented only from one source at a time. Switch of source t New text can be presented from only one source at a time. Switching the sour
o be presented takes place at suitable places in the text, such as end of phrase ce to be presented takes place at suitable places in the text, such as the end o
, end of sentence, line separator and inactivity. Therefore, the time to switch f a phrase, the end of a sentence, or a Line Separator, or upon detecting inacti
to present waiting text from other sources may become long and will vary and dep vity.
end on the actions of the currently presented source. Therefore, the time to switch to present waiting text from other sources may gr
ow long, and it will vary and depend on the actions of the currently presented s
ource.
</t> </t>
<t> <t>
Erasure can only be done up to the latest source switch. If a user tries to Erasure can only be done up to the latest source switch. If a user tries to
erase more text, the erasing actions will be presented as letter X after the lab erase more text, the erasing actions will be presented as a letter "X" after the
el. label.
</t> </t>
<t> <t>
Text loss because of network errors may hit the label between entries from different parties, causing risk for misunderstanding from which source a piece of text is. Text loss because of network errors may hit the label between entries from different parties, causing the risk of a misunderstanding regarding which sourc e provided a piece of text.
</t> </t>
<t> <t>
These facts make it strongly RECOMMENDED implementing multiparty awarenes s in RTT endpoints. The use of the mixing method for multiparty-unaware endpoint s should be left for use with endpoints which are impossible to upgrade to becom e multiparty-aware. Because of these facts, it is strongly <bcp14>RECOMMENDED</bcp14> that m ultiparty awareness be implemented in real-time text endpoints. The use of the m ixing method for multiparty-unaware endpoints should be left for use with endpoi nts that are impossible to upgrade to become multiparty aware.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Example views of presentation on multiparty-unaware endpoints</n ame> <name>Example Views of Presentation on Multiparty-Unaware Endpoints</n ame>
<t> <t>
The following pictures are examples of the view on a participant's display for t The following pictures are examples of the view on a participant's display for t
he multiparty-unaware case. he multiparty-unaware case.</t>
</t>
<t/>
<artwork name="" type="" align="left" alt=""><![CDATA[
_________________________________________________ <t><xref target="fig3"/> shows how a coordinated column view <bcp14>MAY</bcp14>
| Conference | Alice | be presented on Alice's device in a view with two columns. The mixer inserts lab
|________________________|_________________________| els to show how the sources alternate in the column with received text. The mixe
| |I will arrive by TGV. | r alternates between the sources at suitable points in the text exchange so that
|[Bob]:My flight is to |Convenient to the main | text entries from each party can be conveniently read.</t>
|Orly. |station. |
|[Eve]:Hi all, can we | | <figure anchor="fig3" title="Alice, Who Has a Conference-Unaware Clien
|plan for the seminar. | | t, Is Receiving the Multiparty Real-Time Text in a Single Stream">
| | | <artwork name="" type="" align="left" alt=""><![CDATA[
|[Bob]:Eve, will you do | | ___________________________________________________
|your presentation on | | | Conference | Alice |
|Friday? | | |_________________________|_________________________|
|[Eve]:Yes, Friday at 10.| | | |I will arrive by TGV. |
|[Bob]: Fine, wo |We need to meet befo | |[Bob]: My flight is to |Convenient to the main |
|________________________|_________________________| |Orly. |station. |
|[Eve]: Hi all, can we | |
|plan for the seminar. | |
| | |
|[Bob]: Eve, will you do | |
|your presentation on | |
|Friday? | |
|[Eve]: Yes, Friday at 10.| |
|[Bob]: Fine, wo |We need to meet befo |
|_________________________|_________________________|
]]></artwork> ]]></artwork>
<t>Figure 5: Alice who has a conference-unaware client is receiving th </figure>
e multiparty real-time text in a single-stream.</t>
<t> This figure shows how a coordinated column view MAY be presente <t>In <xref target="fig4"/>, there is a tradition in receiving applicatio
d on Alice's device in a view with two-columns. The mixer inserts labels to show ns to include a label showing the source of the text, here shown with parenthese
how the sources alternate in the column with received text. The mixer alternate s "()". The mixer also inserts source labels for the multiparty call participant
s between the sources at suitable points in the text exchange so that text entri s, here shown with brackets "[]".</t>
es from each party can be conveniently read.</t> <figure anchor="fig4" title="An Example of a View of the Multiparty-Un
aware Presentation in Chat Style, Where Alice Is the Local User">
<artwork name="" type="" align="left" alt=""><![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
_________________________________________________ _________________________________________________
| |^| | |^|
|(Alice) Hi, Alice here. |-| |(Alice) Hi, Alice here. |-|
| | | | | |
|(mix)[Bob)] Bob as well. | | |(mix)[Bob] Bob as well. | |
| | | | | |
|[Eve] Hi, this is Eve, calling from Paris | | |[Eve] Hi, this is Eve, calling from Paris | |
| I thought you should be here. | | | I thought you should be here. | |
| | | | | |
|(Alice) I am coming on Thursday, my | | |(Alice) I am coming on Thursday, my | |
| performance is not until Friday morning.| | | performance is not until Friday morning.| |
| | | | | |
|(mix)[Bob] And I on Wednesday evening. | | |(mix)[Bob] And I on Wednesday evening. | |
| | | | | |
|[Eve] we can have dinner and then walk | | |[Eve] we can have dinner and then walk | |
| | | | | |
|[Eve] But I need to be back to | | |[Eve] But I need to be back to | |
| the hotel by 11 because I need | | | the hotel by 11 because I need | |
| |-| | |-|
|______________________________________________|v| |______________________________________________|v|
| of course, I underst | | of course, I underst |
|________________________________________________| |________________________________________________|
]]></artwork> ]]></artwork>
<t>Figure 6: An example of a view of the multiparty-unaware presentati </figure>
on in chat style. Alice is the local user.
</t>
<t>
In this view, there is a tradition in receiving applications to
include a label showing the source of the text, here shown with parenthesis "()
". The mixer also inserts source labels for the multiparty call participants, he
re shown with brackets "[]".
</t>
</section> </section>
</section> </section>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Relation to Conference Control</name> <name>Relationship to Conference Control</name>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Use with SIP centralized conferencing framework</name> <name>Use with SIP Centralized Conferencing Framework</name>
<t> <t>
The Session Initiation Protocol (SIP) conferencing framework, mainly specified in <xref target="RFC4353" format="default"/>, <xref target="RFC4579" format="de fault"/> and <xref target="RFC4575" format="default"/> is suitable for coordinat ing sessions including multiparty RTT. The RTT stream between the mixer and a pa rticipant is one and the same during the conference. Participants get announced by notifications when participants are joining or leaving, and further user info rmation may be provided. The SSRC of the text to expect from joined users MAY be included in a notification. The notifications MAY be used both for security pur poses and for translation to a label for presentation to other users. The Session Initiation Protocol (SIP) conferencing framework, mainly specified in <xref target="RFC4353" format="default"/>, <xref target="RFC4579" format="de fault"/>, and <xref target="RFC4575" format="default"/>, is suitable for coordin ating sessions, including multiparty real-time text. The real-time text stream b etween the mixer and a participant is one and the same during the conference. Pa rticipants get announced by notifications when participants are joining or leavi ng, and further user information may be provided. The SSRC of the text to expect from joined users <bcp14>MAY</bcp14> be included in a notification. The notific ations <bcp14>MAY</bcp14> be used for both security purposes and translation to a label for presentation to other users.
</t> </t>
</section> </section>
<section anchor="confcontrol" numbered="true" toc="default" ><name>Confe rence control</name> <section anchor="confcontrol" numbered="true" toc="default" ><name>Confe rence Control</name>
<t> <t>
In managed conferences, control of the real-time text media SHOULD be provided in the same way as other for media, e.g., for muting and unmuting by th e direction attributes in SDP <xref target="RFC8866"/>. In managed conferences, control of the real-time text media <bcp14>SHO ULD</bcp14> be provided in the same way as for other media, e.g., for muting and unmuting by the direction attributes in SDP <xref target="RFC8866"/>.
</t> </t>
<t> <t>
Note that floor control functions may be of value for RTT users as wel l as for users of other media in a conference. Note that floor control functions may be of value for real-time text u sers as well as for users of other media in a conference.
</t> </t>
</section> </section>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Gateway Considerations</name> <name>Gateway Considerations</name>
<section numbered="true" toc="default"> <t>
<name>Gateway considerations with Textphones</name> Multiparty real-time text sessions may involve gateways of different kinds. G
<t> ateways involved in setting up sessions <bcp14>SHALL</bcp14> correctly reflect t
multiparty RTT sessions may involve gateways of different kinds. Gateways inv he multiparty capability or unawareness of the combination of the gateway and th
olved in setting up sessions SHALL correctly reflect the multiparty capability o e remote endpoint beyond the gateway.
r unawareness of the combination of the gateway and the remote endpoint beyond t
he gateway.
</t> </t>
<section numbered="true" toc="default">
<name>Gateway Considerations with Textphones</name>
<t> <t>
One case that may occur is a gateway to Public Switched Telephone Network (PS TN) for communication with textphones (e.g., TTYs). Textphones are limited devic es with no multiparty awareness, and it SHOULD therefore be suitable for the gat eway to not indicate multiparty awareness for that case. Another solution is tha t the gateway indicates multiparty capability towards the mixer, and includes th e multiparty mixer function for multiparty-unaware endpoints itself. This soluti on makes it possible to adapt to the functional limitations of the textphone. One case that may occur is a gateway to the Public Switched Telephone Network (PSTN) for communication with textphones (e.g., TTYs). Textphones are limited d evices with no multiparty awareness, and it <bcp14>SHOULD</bcp14> therefore be a ppropriate for the gateway to not indicate multiparty awareness for that case. A nother solution is that the gateway indicates multiparty capability towards the mixer and includes the multiparty mixer function for multiparty-unaware endpoint s itself. This solution makes it possible to adapt to the functional limitations of the textphone.
</t> </t>
<t> <t>
More information on gateways to textphones is found in <xref target="RFC5194" format="default"/> More information on gateways to textphones is found in <xref target="RFC5194" format="default"/>.
</t> </t>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Gateway considerations with WebRTC</name> <name>Gateway Considerations with WebRTC</name>
<t> <t>
Gateway operation to real-time text in WebRTC may also be required. In Gateway operation between RTP-mixer-based multiparty real-time text an
WebRTC, RTT is specified in <xref target="RFC8865" format="default"/>. d WebRTC-based real-time text may also be required. Real-time text transport in
WebRTC is specified in <xref target="RFC8865" format="default"/>.
</t> </t>
<t> <t>
A multiparty bridge may have functionality for communicating by RTT both in RTP streams with RTT and WebRTC T.140 data channels. Other configurations may co nsist of a multiparty bridge with either technology for RTT transport and a sepa rate gateway for conversion of the text communication streams between RTP and T. 140 data channel. A multiparty bridge may have functionality for communicating via real-time t ext in both (1) RTP streams with real-time text and (2) WebRTC T.140 data channe ls. Other configurations may consist of a multiparty bridge with either technolo gy for real-time text transport and a separate gateway for conversion of the tex t communication streams between RTP and T.140 data channels.
</t> </t>
<t> <t>
In WebRTC, it is assumed that for a multiparty session, one T.140 data channel i s established for each source from a gateway or bridge to each participant. Each participant also has a data channel with a two-way connection with the gateway or bridge. In WebRTC, it is assumed that for a multiparty session, one T.140 data channel i s established for each source from a gateway or bridge to each participant. Each participant also has a data channel with a two-way connection with the gateway or bridge.
</t> </t>
<t> <t>
The T.140 data channel used both ways is for text from the WebRTC user and fro m the bridge or gateway itself to the WebRTC user. The label parameter of this T .140 data channel is used as the NAME field in RTCP to participants on the RTP s ide. The other T.140 data channels are only for text from other participants to the WebRTC user. A T.140 data channel used for two-way communication is for text from the WebRT C user and from the bridge or gateway itself to the WebRTC user. The label param eter of this T.140 data channel is used as the NAME field in RTCP to participant s on the RTP side. The other T.140 data channels are only for text from other pa rticipants to the WebRTC user.
</t> </t>
<t> <t>
When a new participant has entered the session with RTP transport of RTT, a ne w T.140 channel SHOULD be established to WebRTC users with the label parameter c omposed of information from the NAME field in RTCP on the RTP side. When a new participant has entered the session with RTP transport of real-time text, a new T.140 data channel <bcp14>SHOULD</bcp14> be established to WebRTC u sers with the label parameter composed of information from the NAME field in RTC P on the RTP side.
</t> </t>
<t> <t>
When a new participant has entered the multiparty session with RTT transport i n a WebRTC T.140 data channel, the new participant SHOULD be announced by a noti fication to RTP users. The label parameter from the WebRTC side SHOULD be used a s the NAME RTCP field on the RTP side, or other available session information. When a new participant has entered the multiparty session with real-time text transport in a WebRTC T.140 data channel, the new participant <bcp14>SHOULD</bcp 14> be announced by a notification to RTP users. The label parameter from the We bRTC side or other suitable information from the session or stream establishment procedure <bcp14>SHOULD</bcp14> be used to compose the NAME RTCP field on the R TP side.
</t> </t>
<t> <t>
When a participant on the RTP side is disconnected from the multiparty session , the corresponding T.140 data channel(s) SHOULD be closed. When a participant on the RTP side is disconnected from the multiparty session , the corresponding T.140 data channel(s) <bcp14>SHOULD</bcp14> be closed.
</t> </t>
<t> <t>
When a WebRTC user of T.140 data channels disconnects from the mixer, the corr esponding RTP streams or sources in an RTP-mixed stream SHOULD be closed. When a WebRTC user of T.140 data channels disconnects from the mixer, the corr esponding RTP streams or sources in an RTP-mixed stream <bcp14>SHOULD</bcp14> be closed.
</t> </t>
<t> <t>
T.140 data channels MAY be opened and closed by negotiation or renegotiation o T.140 data channels <bcp14>MAY</bcp14> be opened and closed by negotiation or
f the session or by any other valid means as specified in section 1 of <xref tar renegotiation of the session, or by any other valid means, as specified in
get="RFC8865" format="default"/>. <xref target="RFC8865" sectionFormat="of" section="1"/>.
</t> </t>
</section> </section>
</section> </section>
<section numbered="true" toc="default"> <section numbered="true" toc="default">
<name>Updates to RFC 4103</name> <name>Updates to RFC 4103</name>
<t> <t>
This document updates <xref target="RFC4103" format="default"/> by introducing an SDP media attribute "rtt-mixer" for negotiation of multiparty-mixing capabil ity with the <xref target="RFC4103" format="default"/> format, and by specifying the rules for packets when multiparty capability is negotiated and in use. This document updates <xref target="RFC4103" format="default"/> by introducing an SDP media attribute, "rtt-mixer", for negotiation of multiparty-mixing capab ility with the format described in <xref target="RFC4103" format="default"/> and by specifying the rules for packets when multiparty capability is negotiated an d in use.
</t> </t>
</section> </section>
<section anchor="congest" numbered="true" toc="default"> <section anchor="congest" numbered="true" toc="default">
<name>Congestion considerations</name> <name>Congestion Considerations</name>
<t> <t>
The congestion considerations and recommended actions from <xref target="RFC4103 " format="default"/> are also valid in multiparty situations. The congestion considerations and recommended actions provided in <xref target=" RFC4103" format="default"/> are also valid in multiparty situations.
</t> </t>
<t> <t>
The time values SHALL then be applied per source of text sent to a recei ver. The time values <bcp14>SHALL</bcp14> then be applied per source of text sent to a receiver.
</t> </t>
<t> <t>
If the very unlikely situation appears that many participants in a confe In the very unlikely event that many participants in a conference send t
rence send text simultaneously for a long period, a delay may build up for prese ext simultaneously for a long period of time, a delay may build up for the prese
ntation of text at the receivers if the limitation in characters per second ("cp ntation of text at the receivers if the limitation in characters per second ("cp
s") to be transmitted to the participants is exceeded. More delay than 7 seconds s") to be transmitted to the participants is exceeded. A delay of more than 15 s
can cause confusion in the session. It is therefore RECOMMENDED that an RTP-mix econds can cause confusion in the session. It is therefore <bcp14>RECOMMENDED</b
er-based mixer discards such text causing excessive delays and inserts a general cp14> that an RTP mixer discard such text causing excessive delays and insert a
indication of possible text loss <xref target="T140ad1" format="default"/> in t general indication of possible text loss <xref target="T140ad1" format="default"
he session. If the main text contributor is indicated in any way, the mixer MAY /> in the session.
avoid deleting text from that participant. It should however be noted that human If the main text contributor is indicated in any way, the mixer <bcp14>MAY</bcp
creation of text normally contains pauses, when the transmission can catch up, 14> avoid deleting text from that participant. It should, however, be noted that
so that the transmission overload situations are expected to be very rare. human creation of text normally contains pauses, when the transmission can catc
h up, so that transmission-overload situations are expected to be very rare.
</t> </t>
</section> </section>
<section anchor="IANAConsiderations" numbered="true" toc="default"> <section anchor="IANAConsiderations" numbered="true" toc="default">
<name>IANA Considerations</name> <name>IANA Considerations</name>
<section anchor="reg-rtt-mix" numbered="true" toc="default"> <section anchor="reg-rtt-mix" numbered="true" toc="default">
<name>Registration of the "rtt-mixer" SDP media attribute</name> <name>Registration of the "rtt-mixer" SDP Media Attribute</name>
<t>
[RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
RFC number of this document.]
</t>
<t> <t>
IANA is asked to register the new SDP attribute "rtt-mixer". IANA has registered the new SDP attribute "rtt-mixer".
</t> </t>
<dl newline="false"> <dl newline="false">
<dt>Contact name: <dt>Contact name:
</dt> </dt>
<dd>IESG <dd>IESG
</dd> </dd>
<dt>Contact email: <dt>Contact email:
</dt> </dt>
<dd>iesg@ietf.org <dd>iesg@ietf.org
</dd> </dd>
<dt>Attribute name: <dt>Attribute name:
</dt> </dt>
<dd>rtt-mixer <dd>rtt-mixer
</dd> </dd>
<dt>Attribute semantics: <dt>Attribute semantics:
</dt> </dt>
<dd>See RFCXXXX <xref target="nego1"/> <dd>See RFC 9071, <xref target="nego1"/>
</dd> </dd>
<dt>Attribute value: <dt>Attribute value:
</dt> </dt>
<dd>none <dd>none
</dd> </dd>
<dt>Usage level: <dt>Usage level:
</dt> </dt>
<dd>media <dd>media
</dd> </dd>
<dt>Purpose: <dt>Purpose:
</dt> </dt>
<dd>Indicate support by mixer and endpoint of multiparty mixing for real-time te xt transmission, using a common RTP-stream for transmission of text from a numbe r of sources mixed with one source at a time and the source indicated in a singl e CSRC-list member. <dd>To indicate mixer and endpoint support of multiparty mixing for real-time te xt transmission, using a common RTP stream for transmission of text from a numbe r of sources mixed with one source at a time and where the source is indicated i n a single CSRC-list member.
</dd> </dd>
<dt>Charset Dependent: <dt>Charset Dependent:
</dt> </dt>
<dd>no</dd> <dd>no</dd>
<dt>O/A procedure: <dt>O/A procedures:
</dt> </dt>
<dd>See RFCXXXX <xref target="nego1"/> <dd>See RFC 9071, <xref target="nego1"/>
</dd> </dd>
<dt>Mux Category: <dt>Mux Category:
</dt> </dt>
<dd>normal <dd>normal
</dd> </dd>
<dt>Reference: <dt>Reference:
</dt> </dt>
<dd>RFCXXXX <dd>RFC 9071
</dd> </dd>
</dl> </dl>
</section> </section>
</section> </section>
<section anchor="Security" numbered="true" toc="default"> <section anchor="Security" numbered="true" toc="default">
<name>Security Considerations</name> <name>Security Considerations</name>
<t> <t>
The RTP-mixer model requires the mixer to be allowed to decrypt, pack, and encry The RTP-mixer model requires the mixer to be allowed to decrypt, pack, and encry
pt secured text from the conference participants. Therefore, the mixer needs to pt secured text from conference participants. Therefore, the mixer needs to be t
be trusted rusted
to maintain confidentiality and integrity of the RTT data. This situation is sim to maintain confidentiality and integrity of the real-time text data. This situa
ilar to the situation for handling audio and video media in centralized mixers. tion is similar to the situation for handling audio and video media in centraliz
ed mixers.
</t> </t>
<t> <t>
The requirement to transfer information about the user in RTCP reports in SDES, CNAME, and NAME fields, and in conference notifications, may have privacy concer ns as already stated in RFC 3550 <xref target="RFC3550" format="default"/>, and may be restricted for privacy reasons. When used for creation of readable labels in the presentation, the receiving user will then get a more symbolic label for the source. The requirement to transfer information about the user in RTCP reports in SDES, CNAME, and NAME fields, and in conference notifications, may have privacy concer ns, as already stated in RFC 3550 <xref target="RFC3550" format="default"/>, and may be restricted for privacy reasons. When used for the creation of readable l abels in the presentation, the receiving user will then get a more symbolic labe l for the source.
</t> </t>
<t>
The services available through the RTT mixer may have special interest
for deaf and hard-of-hearing persons. Some users may want to refrain from reveal
ing such characteristics broadly in conferences. The design of the conference sy
stems where the mixer is included MAY need to be made with confidentiality of su
ch characteristics in mind.
</t>
<t>
Participants with malicious intentions may appear and e.g., disturb the
multiparty session by emitting a continuous flow of text. They may also send te
xt that appears to originate from other participants. Counteractions should be t
o require secure signaling, media and authentication, and to provide higher-laye
r conference functions e.g., for blocking, muting, and expelling participants.
</t>
<t>
Participants with malicious intentions may also try to disturb the pres
entation by sending incomplete or malformed control codes. Handling of text from
the different sources by the receivers MUST therefore be well separated so that
the effects of such actions only affect text from the source causing the action
.
</t>
<t>
Care should be taken that if use of the mixer is allowed for users both
with and without security procedures, opens for possible attacks by both unauth
enticated call participants and even eavesdropping and manipulating of content n
on-participants.
</t>
<t>
As already stated in <xref target="security2" format="default"/>, secur
ity in media SHOULD be applied by using DTLS-SRTP <xref target="RFC5764" format=
"default"/> on the media level.
</t>
<t>Further security considerations specific for this application are sp
ecified in <xref target="security2" format="default"/>.
</t>
</section>
<section numbered="true" toc="default">
<name>Change history</name>
<t>
[RFC Editor: Please remove this section prior to publication.]
</t>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-20
</name>
<t>
Inclusion of edits as respone to a comment by Benjamin Kaduk i
n section 3.16.3 to make the recovery procedure generic.
</t>
<t>
Added persons to the acknowledgements and moved acknowledgemen
ts to last in the document.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-19
</name>
<t> <t>
Edits because of comments in a review by Francesca Palombini. The services available through the real-time text mixer may be of spec
</t> ial interest to deaf and hard-of-hearing individuals. Some users may want to ref
<t> rain from revealing such characteristics broadly in conferences. Conference syst
Edits because of comments from Benjamin Kaduk. ems where the mixer is included <bcp14>MAY</bcp14> need to be designed with the
</t> confidentiality of such characteristics in mind.
<t>
Proposed to not change anything because of Robert Wilton's comm
ents.
</t>
<t>
Two added sentences in the security section to meet comments by
Roman Danyliw.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-18
</name>
<t>
Edits of nits as proposed in a review by Lars Eggert.
</t>
<t>
Edits as response to review by Martin Duke.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17
</name>
<t>
Actions on Gen-ART review comments.
</t>
<t>
Actions on SecDir review comments.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16
</name>
<t>
Improvements in the offer/answer considerations section by addi
ng subsections for each phase in the negotiation as requested by IANA expert rev
iew.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15
</name>
<t>
Actions on review comments from Jurgen Schonwalder:
</t>
<t>
A bit more about congestion situations and that they are expect
ed to be very rare.
</t>
<t>
Explanation of differences in security between the conference-a
ware and the conference-unaware case added in security section.
</t>
<t>
Presentation examples with suource labels made
less confusing, and explained.
</t>
<t>
Reference to T.140 inserted at first mentioning
of T.140.
</t>
<t>
Reference to RFC 8825 inserted to explain WebRT
C
</t>
<t>
Nit in wording in terminology s
ection adjusted.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14
</name>
<t>
Changes from comments by Murray Kucherawy during AD review.
</t>
<t>
Many SHOULD in section 4.2 on multiparty-unaware mixing changed
to SHALL, and the whole section instead
specified to be optional depending on the application.
</t>
<t>
Some SHOULD in section 3 either explained or changed to SHALL.
</t>
<t>
In order to have explainable conditions behind SHOULDs, the tra
nsmission interval in 3.4 is changed to as soon as text is available as a main p
rinciple. The call participants send with 300 ms interval so that will create re
alistic load conditions anyway.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13
</name>
<t>
Changed year to 2021.
</t>
<t>
Changed reference to draft on RTT in WebRTC to recently publish
ed RFC 8865.
</t>
<t>
Changed label brackets in example from "[]" to "()" to avoid ni
ts comment.
</t>
<t>
Changed reference "RFC 4566" to recently published "RFC 8866"
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12</n
ame>
<t>
Changes according to responses on comments from Brian Rosen in
Avtcore list on 2020-12-05 and -06.
</t>
<t>
Changes according to responses to comments by Bernard Aboba in
avtcore list 2020-12-06.
</t>
<t>
Introduction of an optiona RTP multi-stream mix
ing method for further study as proposed by Bernard Aboba.
</t>
<t>
Changes clarifying how to open and close T.140 data channels in
cluded in 6.2 after comments by Lorenzo Miniero.
</t>
<t>
Changes to satisfy nits check. Some "not" changed to "NOT" in n
ormative wording combinations. Some lower case normative words changed to upper
case. A normative reference deleted from the abstract. Two informative documents
moved from normative references to informative references.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11</n
ame>
<t>
Timestamps and timestamp offsets added to the packet examples i
n section 3.23, and the description corrected.
</t>
<t>
A number of minor corrections added in sections 3.10 - 3.23.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10</n
ame>
<t>
The packet composition was modified for interleaving packets fr
om different sources.
</t>
<t>
The packet reception was modified for the new interleaving meth
od.
</t>
<t>
The packet sequence examples was adjusted for the new interleav
ing method.
</t>
<t>
Modifications according to responses to Brian Rosen of 2020-11-
03
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09</n
ame>
<t>
Changed name on the SDP media attribute to "rtt-mixer"
</t>
<t>
Restructure of section 2 for balance between aware and unaware cases
.
</t>
<t>
Moved conference control to own section.
</t>
<t>
Improved clarification of recovery and loss in the packet sequence e
xample.
</t>
<t>
A number of editorial corrections and improvements.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08</n
ame>
<t>
Deleted the method requiring a new packet format "text/rex" because
of the longer standardization and implementation period it needs.
</t>
<t>
Focus on use of RFC 4103 text/red format with shorter transmission i
nterval, and source indicated in CSRC.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07</n
ame>
<t>
Added a method based on the "text/red" format and single source per
packet, negotiated by the "rtt-mixer" SDP attribute.
</t>
<t>
Added reasoning and recommendation about indication of loss.
</t>
<t>
The highest number of sources in one packet is 15, not 16. Changed.
</t>
<t>
Added in information on update to RFC 4103 that RFC 4103 explicitly al
lows addition of FEC method. The redundancy is a kind of forward error correctio
n.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06</n
ame>
<t>
Improved definitions list format.
</t> </t>
<t> <t>
The format of the media subtype parameters is made to match the requ irements. Participants with malicious intentions may appear and, for example, di srupt the multiparty session by emitting a continuous flow of text. They may als o send text that appears to originate from other participants. Countermeasures s hould include requiring secure signaling, media, and authentication, and providi ng higher-layer conference functions, e.g., for blocking, muting, and expelling participants.
</t> </t>
<t> <t>
The mapping of media subtype parameters to SDP is included. Participants with malicious intentions may also try to disrupt the pre sentation by sending incomplete or malformed control codes. Handling of text fro m the different sources by the receivers <bcp14>MUST</bcp14> therefore be well s eparated so that the effects of such actions only affect text from the source ca using the action.
</t> </t>
<t> <t>
The "cps" parameter belongs to the t140 subtype and does not need to Care should be taken to avoid the possibility of attacks by
be registered here. unauthenticated call participants, and even eavesdropping and manipulation of co
ntent by non-participants, if the use of the mixer is permitted for users both w
ith and without security procedures.
</t> </t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05</n
ame>
<t> <t>
nomenclature and editorial improvements As already stated in <xref target="security2" format="default"/>, secu rity in media <bcp14>SHOULD</bcp14> be applied by using DTLS-SRTP <xref target=" RFC5764" format="default"/> at the media level.
</t> </t>
<t> <t>Further security considerations specific to this application are sp
"this document" used consistently to refer to this document. ecified in <xref target="security2" format="default"/>.
</t> </t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04</n
ame>
<t>
'Redundancy header' renamed to 'data header'.
</t>
<t>
More clarifications added.
</t>
<t>
Language and figure number corrections.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03</n
ame>
<t>
Mention possible need to mute and raise hands as for other media. ---done ----
</t>
<t>
Make sure that use in two-party calls is also possible and explained. - may ne
ed more wording -
</t>
<t>
Clarify the RTT is often used together with other media. --done--
</t>
<t>
Tell that text mixing is N-1. A users own text is not received in the mix. -do
ne-
</t>
<t>
In 3. correct the interval to:
A "text/rex" transmitter SHOULD send packets distributed in time as long as th
ere is something (new or redundant T140blocks) to transmit. The maximum transm
ission interval SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a r
eceiver as soon as new text to that receiver is available, as long as the time
after the latest sent packet to the same receiver is more than 150 ms, and also
the maximum character rate to the receiver is not exceeded. The intention is to
keep the latency low while keeping a good protection against text loss in bursty
packet loss conditions. -done-
</t>
<t>
In 1.3 say that the format is used both ways. -done-
</t>
<t>
In 13.1 change presentation area to presentation field so that reader does not
think it shall be totally separated. -done-
</t>
<t>
In Performance and intro, tell the performance in number of simultaneous sendi
ng users and introduced delay 16, 150 vs requirements 5 vs 500. -done --
</t>
<t>
Clarify redundancy level per connection. -done-
</t>
<t>
Timestamp also for the last data header. To make it possible for all text to h
ave time offset as for transmission from the source. Make that header equal to t
he others. -done-
</t>
<t>
Mixer always use the CSRC list, even for its own BOM. -done-
</t>
<t>
Combine all talk about transmission interval (300 ms vs when text has arrived)
in section 3 in one paragraph or close to each other. -done-
</t>
<t>
Documents the goal of good performance with low delay for 5 simultaneous typer
s in the introduction. -done-
</t>
<t>
Describe better that only primary text shall be sent on to receivers. Redundan
cy and loss must be resolved by the mixer. -done-
</t>
</section>
<section numbered="true" toc="default">
<name>Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02</nam
e>
<t>
SDP and better description and visibility of security by OSRTP RFC 8634 needed.
</t>
<t>
The description of gatewaying to WebRTC extended.
</t>
<t>
The description of the data header in the packet is improved.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes to draft-ietf-avtcore-multi-party-rtt-mix-01</name>
<t>
2,5,6 More efficient format "text/rex" introduced and attribute a
=rtt-mix deleted.
</t>
<t>
3. Brief about use of OSRTP for security included- More needed.
</t>
<t>
4. Brief motivation for the solution
and why not rtp-translator is used added to intro.
</t>
<t>
7. More limitations for the multipart
y-unaware mixing method inserted.
</t>
<t>
8. Updates to RFC 4102 and 4103 more clearly expressed.
</t>
<t>
9. Gateway to WebRTC started. More needed.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 to
draft-ietf-avtcore-multi-party-rtt-mix-00</name>
<t>
Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00
</t>
<t>
Replaced CDATA in IANA registration table with better coding.
</t>
<t>
Converted to xml2rfc version 3.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 to
-03</name>
<t>
Changed company and e-mail of the author.
</t>
<t>
Changed title to "RTP-mixer formatting of multi-party Real-time t
ext"
to better match contents.
</t>
<t>
Check and modification where needed of use of RFC 2119 words SHAL
L etc.
</t>
<t>
More about the CC value in sections on transmitters and receivers
so that 1-to-1 sessions do not use the mixer format.
</t>
<t>
Enhanced section on presentation for multiparty-unaware endpoints
</t>
<t>
A paragraph recommending cps=150 inserted in the performance sect
ion.
</t>
</section>
<section numbered="true" toc="default">
<name>Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 to
-02</name>
<t>
In Abstract and 1. Introduction: Introduced wording about regulatory requirement
s.
</t>
<t>
In section 5: The transmission interval is decreased to 100 ms when there is tex
t from more than one source to transmit.
</t>
<t>
In section 11 about SDP negotiation, a SHOULD-requirement is introduced that the
mixer should make a mix for multiparty-unaware endpoints if the negotiation is
not successful. And a reference to a later chapter about it.
</t>
<t>
The presentation considerations chapter 14 is extended with more information abo
ut presentation on multiparty-aware endpoints, and a new section on the multipar
ty-unaware mixing with low functionality but SHOULD be implemented in mixers. Pr
esentation examples are added.
</t>
<t>
A short chapter 15 on gateway considerations is introduced.
</t>
<t>
Clarification about the text/t140 format included in chapter 10.
</t>
<t>
This sentence added to the chapter 10 about use without redundancy. "The text/re
d format SHOULD be used unless some other protection against packet loss is util
ized, for example a reliable network or transport."
</t>
<t>
Note about deviation from RFC 2198 added in chapter 4.
</t>
<t>
In chapter 9. "Use with SIP centralized conferencing framework" the following n
ote is inserted:
Note: The CSRC-list in an RTP packet only includes participants whose text is in
cluded in one or more text blocks. It is not the same as the list of participant
s in a conference. With audio and video media, the CSRC-list would often contain
all participants who are not muted whereas text participants that don't type ar
e completely silent and so don't show up in RTP packet CSRC-lists.
</t>
<t>
</t>
</section>
<section numbered="true" toc="default">
<name>Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 to
-01</name>
<t>
Editorial cleanup.
</t>
<t>
Changed capability indication from fmtp-parameter to SDP attribute "rt
t-mix".
</t>
<t>
Swapped order of redundancy elements in the example to match reality.
</t>
<t>
Increased the SDP negotiation section
</t>
</section>
</section> </section>
</middle> </middle>
<back> <back>
<references> <references>
<name>References</name> <name>References</name>
<references> <references>
<name>Normative References</name> <name>Normative References</name>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.2119.xml"/> FC.2119.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.4102.xml"/> FC.4102.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.4103.xml"/> FC.4103.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.3550.xml"/> FC.3550.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxm <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
l/reference.RFC.5630.xml"/> FC.5630.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.5764.xml"/> FC.5764.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refere <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
nce.RFC.6263.xml"/> FC.6263.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refere <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
nce.RFC.7675.xml"/> FC.7675.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refere <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
nce.RFC.8174.xml"/> FC.8174.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refere <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
nce.RFC.8865.xml"/> FC.8865.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refere <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
nce.RFC.8866.xml"/> FC.8866.xml"/>
<reference anchor="T140" target="https://www.itu.int/rec/T-REC-T.140-1998
02-I/en"> <reference anchor="T140" target="https://www.itu.int/rec/T-REC-T.140-199
802-I/en">
<front> <front>
<title>Recommendation ITU-T T.140 (02/1998), Protocol for <title>Protocol for multimedia application text conversation</title>
multimedia application text conversation</title>
<author> <author>
<organization>ITU-T</organization> <organization>ITU-T</organization>
</author> </author>
<date year="1998" month="February"/> <date year="1998" month="February"/>
</front> </front>
<seriesInfo name="ITU-T Recommendation" value="T.140"/>
</reference> </reference>
<reference anchor="T140ad1" target="https://www.itu.int/rec/T-REC-T.140- 200002-I!Add1/en"> <reference anchor="T140ad1" target="https://www.itu.int/rec/T-REC-T.140- 200002-I!Add1/en">
<front> <front>
<title>Recommendation ITU-T.140 Addendum 1 - (02/2000), Protocol for <title>Recommendation T.140 Addendum</title>
multimedia application text conversation</title>
<author> <author>
<organization>ITU-T</organization> <organization>ITU-T</organization>
</author> </author>
<date year="2000" month="February"/> <date year="2000" month="February"/>
</front> </front>
</reference> </reference>
</references> <reference anchor="ISO6429" target="https://www.iso.org/o
<references> bp/ui/#iso:std:iso-iec:6429:ed-3:v1:en">
<front>
<title>Information technology - Control functions for coded characte
r sets</title>
<author>
<organization>ISO/IEC</organization>
</author>
<date year="1992" month="December"/>
</front>
<seriesInfo name="ISO/IEC" value="ISO/IEC 6429:1992"/>
</reference>
<reference anchor="ECMA-48" target="https://www.ecma-international.org/publi
cations-and-standards/standards/ecma-48/">
<front>
<title>ECMA-48: Control functions for coded character sets</title>
<author>
<organization>Ecma International</organization>
</author>
<date year="1991" month="June"/>
</front>
<refcontent>5th edition</refcontent>
</reference>
</references>
<references>
<name>Informative References</name> <name>Informative References</name>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.4353.xml"/> FC.4353.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.4579.xml"/> FC.4579.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.4575.xml"/> FC.4575.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
ence.RFC.5194.xml"/> FC.5194.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxm <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
l/reference.RFC.7667.xml"/> FC.7667.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxm <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
l/reference.RFC.8643.xml"/> FC.8643.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxm <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
l/reference.RFC.8723.xml"/> FC.8723.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxm <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.R
l/reference.RFC.8825.xml"/> FC.8825.xml"/>
</references> </references>
</references> </references>
<section anchor="Acknowledgements" numbered="false" toc="default"> <section anchor="Acknowledgements" numbered="false" toc="default">
<name>Acknowledgements</name> <name>Acknowledgements</name>
<t>The author want to thank the following persons for support, review <t>The author wants to thank the following persons for support, revi
s and valuable comments: Bernard Aboba, Amanda Baber, Roman Danyliw, Spencer Daw ews, and valuable comments: <contact fullname="Bernard Aboba"/>, <contact fullna
kins, Martin Duke, Lars Eggert, James Hamlin, Benjamin Kaduk, Murray Kucherawy, me="Amanda Baber"/>, <contact fullname="Roman Danyliw"/>, <contact fullname="Spe
Paul Kyziwat, Jonathan Lennox, Lorenzo Miniero, Dan Mongrain, Francesca Palombin ncer Dawkins"/>, <contact fullname="Martin Duke"/>, <contact fullname="Lars Egge
i, Colin Perkins, Brian Rosen, Juergen Schoenwaelder, Rich Salz, Robert Wilton, rt"/>, <contact fullname="James Hamlin"/>, <contact fullname="Benjamin Kaduk"/>,
Dale Worley, Peter Yee and Yong Xin.</t> <contact fullname="Murray Kucherawy"/>, <contact fullname="Paul Kyzivat"/>, <co
ntact fullname="Jonathan Lennox"/>, <contact fullname="Lorenzo Miniero"/>, <cont
act fullname="Dan Mongrain"/>, <contact fullname="Francesca Palombini"/>, <conta
ct fullname="Colin Perkins"/>, <contact fullname="Brian Rosen"/>, <contact fulln
ame="Rich Salz"/>,
<contact fullname="Jürgen Schönwälder"/>, <contact fullname="Robert Wilton"/>, <
contact fullname="Dale Worley"/>, <contact fullname="Yong Xin"/>,
and <contact fullname="Peter Yee"/>.
</t>
</section> </section>
</back> </back>
</rfc> </rfc>
 End of changes. 287 change blocks. 
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