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Internet Engineering Task Force                             G. Hellstrom
Internet-Draft                 Gunnar Hellstrom Accessible Communication
Intended status: Informational                             8 August 2020
Expires: 9 February 2021


           Real-time text solutions for multi-party sessions
          draft-hellstrom-avtcore-multi-party-rtt-solutions-03

Abstract

   This document specifies methods for Real-Time Text (RTT) media
   handling in multi-party calls.  The main discussed transport is to
   carry Real-Time text by the RTP protocol in a time-sampled mode
   according to RFC 4103.  The mechanisms enable the receiving
   application to present the received real-time text media, separated
   per source, in different ways according to user preferences.  Some
   presentation related features are also described explaining suitable
   variations of transmission and presentation of text.

   Call control features are described for the SIP environment.  A
   number of alternative methods for providing the multi-party
   negotiation, transmission and presentation are discussed and a
   recommendation for the main ones is provided.  The main solution for
   SIP based centralized multi-party handling of real-time text is
   achieved through a media control unit coordinating multiple RTP text
   streams into one RTP stream.

   Alternative methods using a single RTP stream and source
   identification inline in the text stream are also described, one of
   them being provided as a lower functionality fallback method for
   endpoints with no multi-party awareness for RTT.

   Bridging methods where the text stream is carried without the
   contents being dealt with in detail by the bridge are also discussed.

   Brief information is also provided for multi-party RTT in the WebRTC
   environment.

   The intention is to provide background for decisions, specification
   and implementation of selected methods.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.





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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  Requirements Language . . . . . . . . . . . . . . . . . .   5
   2.  Centralized conference model  . . . . . . . . . . . . . . . .   5
   3.  Requirements on multi-party RTT . . . . . . . . . . . . . . .   6
     3.1.  General requirements  . . . . . . . . . . . . . . . . . .   6
     3.2.  Performance requirements  . . . . . . . . . . . . . . . .   7
   4.  RTP based solutions . . . . . . . . . . . . . . . . . . . . .   8
     4.1.  Coordination of text RTP streams  . . . . . . . . . . . .   8
       4.1.1.  RTP-based solutions with a central mixer  . . . . . .   8
         4.1.1.1.  RTP Mixer using default RFC 4103 methods  . . . .   8
         4.1.1.2.  RTP Mixer using the default method but decreased
                 transmission interval . . . . . . . . . . . . . . .   9
         4.1.1.3.  RTP Mixer with frequent transmission and indicating
                 sources in CSRC-list  . . . . . . . . . . . . . . .  10
         4.1.1.4.  RTP Mixer using timestamp to identify
                 redundancy  . . . . . . . . . . . . . . . . . . . .  11
         4.1.1.5.  RTP Mixer with multiple primary data in each packet
                 and individual sequence numbers . . . . . . . . . .  12
         4.1.1.6.  RTP Mixer with multiple primary data in each
                 packet  . . . . . . . . . . . . . . . . . . . . . .  13



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         4.1.1.7.  RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy
                 in the packets  . . . . . . . . . . . . . . . . . .  14
         4.1.1.8.  RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy
                 and separate sequence number in the packets . . . .  16
         4.1.1.9.  RTP Mixer indicating participants by a control code
                 in the stream . . . . . . . . . . . . . . . . . . .  18
         4.1.1.10. Mixing for multi-party unaware user agents  . . .  20
       4.1.2.  RTP-based bridging with minor RTT media contents
               reformatting by the bridge  . . . . . . . . . . . . .  21
         4.1.2.1.  RTP Translator sending one RTT stream per
                 participant . . . . . . . . . . . . . . . . . . . .  21
         4.1.2.2.  Distributing packets in an end-to-end encryption
                 structure . . . . . . . . . . . . . . . . . . . . .  24
         4.1.2.3.  Mesh of RTP endpoints . . . . . . . . . . . . . .  25
         4.1.2.4.  Multiple RTP sessions, one for each
                 participant . . . . . . . . . . . . . . . . . . . .  25
   5.  Preferred RTP-based multi-party RTT transport method  . . . .  26
   6.  Session control of RTP-based multi-party RTT sessions . . . .  26
     6.1.  Implicit RTT multi-party capability indication  . . . . .  27
     6.2.  RTT multi-party capability declared by SIP media-tags . .  28
     6.3.  SDP media attribute for RTT multi-party capability
           indication  . . . . . . . . . . . . . . . . . . . . . . .  29
     6.4.  Simplified SDP media attribute for RTT multi-party
           capability indication . . . . . . . . . . . . . . . . . .  31
     6.5.  SDP format parameter for RTT multi-party capability
           indication  . . . . . . . . . . . . . . . . . . . . . . .  31
     6.6.  A text media subtype for support of multi-party rtt . . .  33
     6.7.  Preferred capability declaration method for RTP-based
           transport.  . . . . . . . . . . . . . . . . . . . . . . .  33
     6.8.  Identification of the source of text for RTP-based
           solutions . . . . . . . . . . . . . . . . . . . . . . . .  33
   7.  RTT bridging in WebRTC  . . . . . . . . . . . . . . . . . . .  34
     7.1.  RTT bridging in WebRTC with one data channel per
           source  . . . . . . . . . . . . . . . . . . . . . . . . .  34
     7.2.  RTT bridging in WebRTC with one common data channel . . .  35
     7.3.  Preferred rtt multi-party method for WebRTC . . . . . . .  35
   8.  Presentation of multi-party text  . . . . . . . . . . . . . .  36
     8.1.  Associating identities with text streams  . . . . . . . .  36
     8.2.  Presentation details for multi-party aware endpoints. . .  36
       8.2.1.  Bubble style presentation . . . . . . . . . . . . . .  37
       8.2.2.  Other presentation styles . . . . . . . . . . . . . .  38
   9.  Presentation details for multi-party unaware endpoints. . . .  39
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  39
   11. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  39
   12. Congestion considerations . . . . . . . . . . . . . . . . . .  40
   13. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  40
   14. Change history  . . . . . . . . . . . . . . . . . . . . . . .  40




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     14.1.  Changes to
            draft-hellstrom-avtcore-multi-party-rtt-solutions-03 . .  40
     14.2.  Changes to
            draft-hellstrom-avtcore-multi-party-rtt-solutions-02 . .  40
     14.3.  Changes to
            draft-hellstrom-avtcore-multi-party-rtt-solutions-01 . .  40
     14.4.  Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to
            draft-hellstrom-avtcore-multi-party-rtt-solutions-00 . .  41
     14.5.  Changes from version
            draft-hellstrom-mmusic-multi-party-rtt-01 to -02 . . . .  41
   15. References  . . . . . . . . . . . . . . . . . . . . . . . . .  41
     15.1.  Normative References . . . . . . . . . . . . . . . . . .  41
     15.2.  Informative References . . . . . . . . . . . . . . . . .  42
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  45

1.  Introduction

   Real-time text (RTT) is a medium in real-time conversational
   sessions.  Text entered by participants in a session is transmitted
   in a time-sampled fashion, so that no specific user action is needed
   to cause transmission.  This gives a direct flow of text in the rate
   it is created, that is suitable in a real-time conversational
   setting.  The real-time text medium can be combined with other media
   in multimedia sessions.

   Media from a number of multimedia session participants can be
   combined in a multi-party session.  The present document specifies
   how the real-time text streams can be handled in multi-party
   sessions.  Recommendations are provided for preferred methods.

   The description is mainly focused on the transport level, but also
   describes a few session and presentation level aspects.

   Transport of real-time text is specified in RFC 4103 [RFC4103] RTP
   Payload for text conversation.  It makes use of RFC 3550 [RFC3550]
   Real Time Protocol, for transport.  Robustness against network
   transmission problems is normally achieved through redundant
   transmission based on the principle from RFC 2198 [RFC2198], with one
   primary and two redundant transmission of each text element.  Primary
   and redundant transmissions are combined in packets and described by
   a redundancy header.  This transport is usually used in the SIP
   Session Initiation Protocol RFC 3261 [RFC3261] environment.

   A very brief overview of functions for real-time text handling in
   multi-party sessions is described in RFC 4597 [RFC4597] Conferencing
   Scenarios, sections 4.8 and 4.10.  The present specification builds
   on that description and indicates which protocol mechanisms should be
   used to implement multi-party handling of real-time text.



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   Real-time text can also be transported in the WebRTC environment, by
   using WebRTC data channels according to
   [I-D.ietf-mmusic-t140-usage-data-channel].  Multi-party aspects for
   WebRTC solutions are briefly covered.

1.1.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

2.  Centralized conference model

   In the centralized conference model for SIP, introduced in RFC 4353
   [RFC4353] "A Framework for Conferencing with the Session Initiation
   Protocol (SIP)", one function co-ordinates the communication with
   participants in the multi-party session.  This function also controls
   media mixer functions for the media appearing in the session.  The
   central function is common for control of all media, while the media
   mixers may work differently for each media.

   The central function is called the Focus UA.  Many variants exist for
   setting up sessions including the multipoint control centre.  It is
   not within scope of this description to describe these, but rather
   the media specific handling in the mixer required to handle multi-
   party calls with RTT.

   The main principle for handling real-time text media in a centralized
   conference is that one RTP session for real-time text is established
   including the multipoint media control centre and the participating
   endpoints which are going to have real-time text exchange with the
   others.

   The different possible mechanisms for mixing and transporting RTT
   differs in the way they multiplex the text streams and how they
   identify the sources of the streams.  RFC 7667 [RFC7667] describes a
   number of possible use cases for RTP.  This specification refers to
   different sections of RFC 7667 for further reading of the situations
   caused by the different possible design choices.

   The recommended method for using RTP based RTT in a centralized
   conference model is specified in
   [I-D.ietf-avtcore-multi-party-rtt-mix] based on the recommendations
   in this document.







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   Real-time text can also be transported in the WebRTC environment, by
   using WebRTC data channels according to
   [I-D.ietf-mmusic-t140-usage-data-channel].  Ways to handle multi-
   party calls in that environmnent are also specified.

3.  Requirements on multi-party RTT

3.1.  General requirements

   The following general requirements are placed on multi-party RTT:

      A solution shall be applicable to IMS (3GPP TS 22.173)[TS22173],
      SIP based VoIP and Next Generation Emergency Services (NENA i3
      [NENAi3], ETSI TS 103 479 [TS103479], RFC 6443[RFC6443]).

      The transmission interval for text should not be longer than 500
      milliseconds when there is anything available to send.  Ref ITU-T
      T.140 [T140].

      If text loss is detected or suspected, a missing text marker
      should be inserted in the text stream.  Ref ITU-T T.140 Amendment
      1 [T140ad1].  ETSI EN 301 549 [EN301549]

      The display of text from the members of the conversation shall be
      arranged so that the text from 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 session should be provided.  The text
      should be displayed as soon as it is received.  Ref ITU-T T.140
      [T140]

      Bridges must be multimedia capable (voice, video, text).  Ref NENA
      i3 STA-010.2.  [NENAi3]

      It MUST be possible to use real-time text in conferences both as a
      medium of discussion between individual participants (for example,
      for sidebar discussions in real-time text while listening to the
      main conference audio) and for central support of the conference
      with real-time text interpretation of speech.  Ref (R7) in RFC
      5194.[RFC5194]

      It should be possible to protect RTT contents with usual means for
      privacy and integrity.  Ref RFC 6881 section 16.  [RFC6881]

      Conferencing procedures are documented in RFC 4579 [RFC4579].  Ref
      NENA i3 STA-010.2.[NENAi3]





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      Conferencing applies to any kind of media stream by which users
      may want to communicate.  Ref 3GPP TS 24.147 [TS24147]

      The framework for SIP conferences is specified in RFC 4353
      [RFC4353].  Ref 3GPP TS 24.147 [TS24147]

3.2.  Performance requirements

   The mixer performance requirements can be expressed in one number,
   extracted from the user requirements on real-time text expressed in
   ITU-T F.700, where it is stated that for "good" usability, text
   characters should not be delayed more than 1 second from creation to
   presentation.  For "usable" usability the figure is 2 seconds.  The
   main factor behind these limits is from when taking turns in a
   conversation gets disturbed by a delay of when a response gets
   visible to the receiving part.  If that times get too long, the
   receiving part gets unsure if the previous utterance was well
   perceived and the receiving part maybe prepares for repetition.  This
   is similar to the same effect in voice communication, where the
   usability limit is 400 ms delay.

   Another important factor in a multi-party conference is the
   opportunity for a participant using real-time text to provide timely
   comments and get a chance to enter the discussion if the majority of
   participants use voice in the conference.  A complicating factor when
   stating the requirements is that some transport methods do not cause
   a total delay, but instead an increasing jerkiness when the number of
   simultaneously sending participants is increased.

   It should however be remembered that the expected number of
   participants sending real-time text simultaneously is low.  Just as
   with voice or sign language, the capability of the participants to
   perceive utterances from more than one participant at a time is very
   limited.  Therefore the normal case in multi-party situations is that
   one participant at a time is the main provider of text.  Others might
   usually just provide very brief comments such as "yes" or "no" or
   "may I comment?".  Only at very rare situations two participants
   provide more information simultaneously.

   *  The number of expected simultaneously transmitting users is
      different for different applications.  In all cases, just one
      transmitting user is the normal case.  Two simultaneously
      transmitting participants can occasionally be expected in
      emergency services, relay services, small unmanaged conferences
      and group calls and large managed conferences.  Three
      simultaneously transmitting participants may appear occasionally
      in large unmanaged conferences.  The following can therefore
      express the performance requirement.



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   *  The mean delay of text passing the mixer introduced when only one
      participant is sending text should be kept to a minimum and should
      not be more than 400 ms.

   *  The mean delay of text passing the mixer should not be more than 1
      second during moments when up to three users are sending text
      simultaneously.

   *  For the very rare case that more than three participants send text
      simultaneously, the mixer may take action to limit the introduced
      delay of the text passing the mixer to 7 seconds e.g. by
      discarding text from some participants and instead inserting a
      general warning about possible text loss in the stream.

4.  RTP based solutions

4.1.  Coordination of text RTP streams

   Coordinating and sending text RTP streams in the multi-party session
   can be done in a number of ways.  The most suitable methods are
   specified here with pros and cons.

   A receiving and presenting endpoint MUST separate text from the
   different sources and identify and display them accordingly.

4.1.1.  RTP-based solutions with a central mixer

   A set of solutions can be based on the central RTP mixer.  They are
   described here and a preferred method selected.

4.1.1.1.  RTP Mixer using default RFC 4103 methods

   Without any extra specifications, a mixer would transmit with 300
   milliseconds intervals, and use RFC 4103 [RFC4103] with the default
   redundancy of one original and two redundant transmissions.  The
   source of the text would be indicated by a single member in the CSRC
   list.  Text from different sources cannot be transmitted in the same
   packet.  Therefore, from the time when the mixer sent one piece of
   new text from one source, it will need to transmit that text again
   twice as redundant data, before it can send text from another source.
   The jerkiness = time between transmission of new text is 900 ms.
   This is clearly insufficient.

   Pros:

   Only a capability negotiation method is needed.  No other update of
   standards are needed, just a general remark that traditional RTP-
   mixing is used.



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   Cons:

   Clearly insufficient mixer switching performance.

   A bit complex handling of transmission when there is new text
   available from more than one source.  The mixer needs to send two
   packets more with redundant text from the current source before
   starting to send anything from the other source.

4.1.1.2.  RTP Mixer using the default method but decreased transmission
          interval

   This method makes use of the default RTP-mixing method briefly
   described in Section 4.1.1.1.  The only difference is that the
   transmission interval is decreased to 100 milliseconds when there is
   text from more than one source available for transmission.  The
   jerkiness is 300 ms.  The mean delay with two simultaneously sending
   participants is 250 ms, and with three simultaneously sending
   participants 500 ms.  This is acceptable performance.

   Pros:

   Minor influence on standards

   Can be relatively rapidly be introduced in the intended technical
   environments.

   Can be declared in sdp as the already existing "text/red" format with
   a multi-party attribute for capability negotiation.

   Cons:

   The introduced jerkiness of new text from more than the required
   three simultaneously sending sources is high.

   Slightly higher risk for loss of text at bursty packet loss than for
   the recommended transmission interval (300 ms) for RFC 4103.

   When complete loss of packets occur (beyond recovery), it is not
   possible to deduct from which source text was lost.

   A bit complex handling of transmission when there is new text
   available from more than one source.  The mixer needs to send two
   packets more with redundant text from the current source before
   starting to send anything from the other source.






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4.1.1.3.  RTP Mixer with frequent transmission and indicating sources in
          CSRC-list

   An RTP media mixer combines text from participants into one RTP
   stream, thus all using the same destination address/port combination,
   the same RTP SSRC, and one sequence number series as described in
   Section 7.1 and 7.3 of RTP RFC 3550 [RFC3550] about the Mixer
   function.  This method is also briefly described in RFC 7667, section 
   3.6.1 Media mixing mixer [RFC7667].

   The sources of the text in each RTP packet are identified by the CSRC
   list in the RTP packets, containing the SSRC of the initial sources
   of text.  The order of the CSRC parameters is with the SSRC of the
   source of the primary text first, followed by the SSRC of the first
   level redundancy, and then the second level redundancy.

   The transmission interval should be 100 milliseconds when there is
   text to transmit from more than one source, and otherwise 300 ms.


   The identification of the sources is made through the CSRC fields and
   can be made more readable at the receiver through the RTCP SDES CNAME
   and NAME packets as described in RTP[RFC3550].

   Information provided through the notification according to RFC 4575
   [RFC4575] when the participant joined the conference provides also
   suitable information and a reference to the SSRC.

   A receiving endpoint is supposed to separate text items from the
   different sources and identify and display them accordingly.

   The ordered CSRC lists in the RFC 4103 [RFC4103] packets make it
   possible to recover from loss of one and two packets in sequence and
   assign the recovered text to the right source.  For more loss, a
   marker for possible loss should be inserted or presented.

   The conference server needs to have authority to decrypt the payload
   in the received RTP packets in order to be able to recover text from
   redundant data or insert the missing text marker in the stream, and
   repack the text in new packets.

   Even if the format is very similar to "text/red" of RFC 4103, it
   needs to be declared as a new media subtype, e.g. "text/rex".


   Pros:





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   This method has low overhead and less complexity than the methods in
   Section 4.1.1.1, Section 4.1.1.2, Section 4.1.1.4 and
   Section 4.1.1.6.

   When loss of packets occur, it is possible to recover text from
   redundancy at loss of up to the number of redundancy levels carried
   in the RFC 4103 [RFC4103] stream (normally primary and two redundant
   levels).

   This method can be implemented with most RTP implementations.

   The source switching performance is sufficient for well-behaving
   conference participants.  The jerkiness is 100 ms.

   Cons:

   When more consecutive packet loss than the number of generations of
   redundant data appears, it is not possible to deduct the sources of
   the totally lost data.

   Slightly higher risk for loss of text at bursty packet loss than for
   the recommended transmission interval for RFC 4103.

   Requires a different sub media format, e.g. "text/rex".  This takes a
   long time in standardisation and releases of target technical
   environments.

   The conference server needs to be allowed to decrypt/encrypt the
   packet payload.  This is however normal for media mixers for other
   media.

4.1.1.4.  RTP Mixer using timestamp to identify redundancy

   This method has text only from one source per packet, as the original
   RFC 4103 [RFC4103] specifies.  Packets with text from different
   sources are instead allowed to be merged.  The recovery procedure in
   the receiver will use the RTP timestamp and timestamp offsets in the
   redundancy headers to evaluate if a piece of redundant data should be
   recovered or not in case of packet loss.

   In this method, the transmission interval is 100 milliseconds when
   text from more than one source is available for transmission.

   Pros:

   The format of each packet is equal to what is specified in RFC 4103
   [RFC4103].




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   The source switching performance is sufficient.  Text from five
   participants can be transmitted simultaneously with 500 milliseconds
   interval per source.

   New text from five simultaneous sources can be transmitted within 500
   milliseconds.  This is sufficient.

   Cons:

   The recovery time in case of packet loss is long.  With five
   simultaneously sending participants, it will be 1.5 seconds.

   The recovery procedure is complex and very different from what is
   described in RFC 4103 [RFC4103].

   It is not sure that this change can be regarded to be an update to
   RFC 4103.  It may need a new media subtype.

4.1.1.5.  RTP Mixer with multiple primary data in each packet and
          individual sequence numbers

   This method allows primary as well as redundant text from more than
   one source per packet.  The packet payload contains an ordered set of
   redundant and primary data with the same number of generations of
   redundancy as once agreed in the SDP negotiation.  The data header
   reflects these parts of the payload.  The CSRC list contains one CSRC
   member per source in the payload and in the same order.  An
   individual sequence number per source is included in the data header
   replacing the t140 payload type number that is instead assumed to be
   constant in this format.  This allows an individual extra sequence
   number per source with maximum value 127, suitable for checking for
   which source loss of text appeared when recovery was not possible.

   The data header would contain the following fields:
     0                   1                    2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |F| Source-seq  |  timestamp offset         |   block length    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   Where "Source-seq" is the sequence number per source.

   The maximum number of members in the CSRC-list is 15, and that is
   therefore the maximum number of sources that can be represented in
   each packet provided that all data can be fitted into the size
   allowable in one packet.






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   Transmission is done as soon as there is new text available, but not
   with shorter interval than 150 ms and not longer than 300 ms while
   there is anything to send.

   A new media subtype is needed, e.g. "text/rex".

   This is an SDP offer example for both traditional "text/red"
   and multi-party "text/rex" format:

         m=text 11000 RTP/AVP 101 100 98
         a=rtpmap:98 t140/1000
         a=rtpmap:100 red/1000
         a=rtpmap:101 rex/1000
         a=fmtp:100 98/98/98
         a=fmtp:101 98/98/98

   Pros:

   The source switching performance is good.  Text from 15 participants
   can be transmitted simultaneously.

   New text from 15 simultaneous sources can be transmitted within 300
   milliseconds.  This is good performance.

   When more consecutive packet loss than the number of generations of
   redundant data appears, it is still possible to deduct the sources of
   the totally lost data, when next text from these sources arrive.

   Cons:

   The format of each packet is different from what is specified in RFC
   4103 [RFC4103].

   The processing time in standard organisation will be long.

   A new media subtype is needed, causing a bit complex negotiation.

   The recovery procedure is a bit complex.


4.1.1.6.  RTP Mixer with multiple primary data in each packet

   This method allows primary as well as redundant text from more than
   one source per packet.  The packet payload contains an ordered set of
   redundant and primary data with the same number of generations of
   redundancy as once agreed in the SDP negotiation.  The data header
   reflects these parts of the payload.  The CSRC list contains one CSRC
   member per source in the payload and in the same order.



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   The maximum number of members in the CSRC-list is 15, and that is
   therefore the maximum number of sources that can be represented in
   each packet provided that all data can be fitted into the size
   allowable in one packet.

   Transmission is done as soon as there is new text available, but not
   with shorter interval than 150 ms and not longer than 300 ms while
   there is anything to send.

   A new media subtype is needed, e.g. "text/rex".

   SDP would be the same as in Section 4.1.1.6.

   Pros:

   The source switching performance is good.  Text from 15 participants
   can be transmitted simultaneously.

   New text from 15 simultaneous sources can be transmitted within 150
   milliseconds.  This is good performance.

   Cons:

   The format of each packet is different from what is specified in RFC
   4103 [RFC4103].

   A new media subtype is needed.

   A new media subtype is needed, causing a bit complex negotiation.

   The processing time in standard organisation will be long.

   The recovery procedure is a bit complex [RFC4103].

   When more consecutive packet loss than the number of generations of
   redundant data appears, it is not possible to deduct the sources of
   the totally lost data.

4.1.1.7.  RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy in the
          packets

   This method allows primary data from one source and redundant text
   from other sources in each packet.  The packet payload contains
   primary data in "text/t140" format, and redundant data in RFC 5109
   FEC [RFC5109] format called "text/ulpfec".  That means that the
   redundant data contains the sequence number and the CSRC and other
   characteristics from the RTP header when the data was sent as
   primary.  The redundancy can be sent at a selected number of packets



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   after when it was sent as primary, in order to improve the protection
   against bursty packet loss.  The redundancy level is recommended to
   be the same as in original RFC 4103.

   RFC 4103 says that the protection against loss can be made by other
   methods than plain redundancy, so this method is in line with that
   statement.

   Transmission is done as soon as there is new text available, but not
   with shorter interval than 100 ms and not longer than 300 ms while
   there is anything to send (new or redundant text).

   When more consecutive packet loss than the number of generations of
   redundant data appears, it is not possible to deduct the sources of
   the totally lost data.

   The sdp can indicate the format as "text/red" with "text/ulpfec"
   redundant data in this way. with traditional RFC 4103 with "text/red"
   with "text/t140" as redundant data as a fallback.

   m=text 49170 RTP/AVP 98 101 100 102
   a=rtpmap:98 red/1000
   a=fmtp:98 100/102/102
   a=rtpmap:102 ulpfec/1000
   a=rtpmap:100 t140/1000
   a=rtpmap:101 red/1000
   a=fmtp:101 100/100/100
   a=fmtp:100 cps=200

   The "text/ulpfec" format includes an indication of how far back the
   redundancy belongs, making it possible to cover bursty packet loss
   better than the other formats with short transmission intervals.  For
   real-time text, it is recommended to send three packets between the
   primary and the redundant transmissions of text.  That makes the
   transmission cover between 500 and 1500 ms of bursty packet loss.
   The variation is because of the varying packet interval between many
   and one simultaneously transmitting source.

   The "text/ulpfec" format has a number of parameters.  One is the
   length of the data to be protected which in this case must be the
   whole t140block.

   Pros:

   The source switching performance is good.  Text from 5 participants
   can be transmitted within 500 ms.

   Good recovery from bursty packet loss.



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   The method is based on existing standards.  No new registrations are
   needed.

   Cons:

   When more consecutive packet loss than the number of generations of
   redundant data appears, it is not possible to deduct the sources of
   the totally lost data.

   Even if the switching performance is good, it is not as good as for
   the method called "RTP Mixer with multiple primary data in each
   packet "Section 4.1.1.6.  With more than 5 simultaneously sending
   sources, there will be a noticeable delay of text of over 500 ms,
   with 100 ms added per simultaneous source.  This is however beyond
   the requirements and would be a concern only in congestion
   situations.

   The recovery procedure is a bit complex [RFC5109].

   There is more overhead in terms of extra data and extra packets sent
   than in the other methods.  With the recommended two redundant
   generations of data, each packet will be 36 bytes longer than with
   traditional RFC 4103, and at each pause in transmission five extra
   packets with only redundant data will be sent compared to two extra
   packets for the traditional RFC 4103 case.

4.1.1.8.  RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy and
          separate sequence number in the packets

   This method allows primary data from one source and redundant text
   from other sources in each packet.  The packet payload contains
   primary data in a new "text/t140e" format, and redundant data in RFC
   5109 FEC [RFC5109] format called "text/ulpfec".  That means that the
   redundant data contains the sequence number and the CSRC and other
   characteristics from the RTP header when the data was sent as
   primary.  The redundancy can be sent at a selected number of packets
   after when it was sent as primary, in order to improve the protection
   against bursty packet loss.  The redundancy level is recommended to
   be the same as in original RFC 4103.  The "text/t140e" format
   contains a source-specific sequence number and the t140block.

   RFC 4103 says that the protection against loss can be made by other
   methods than plain redundancy, so this method is in line with that
   statement.

   Transmission is done as soon as there is new text available, but not
   with shorter interval than 100 ms and not longer than 300 ms while
   there is anything to send (new or redundant text).



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   When more consecutive packet loss than the number of generations of
   redundant data appears, it is possible to deduct which sources lost
   data when new data arrives from the sources.  This is done by
   monitoring the received source specific sequence numbers preceding
   the text.

   This is an example of how can indicate the format as "text/red" with
   "text/t140e" as primary and "text/ulpfec" redundant data, with
   traditional RFC 4103 with "text/red" with "text/t140" as redundant
   data as a fallback.

   m=text 49170 RTP/AVP 98 101 100 102 103
   a=rtpmap:98 red/1000
   a=fmtp:98 100/102/102
   a=rtpmap:102 ulpfec/1000
   a=rtpmap:103 t140/1000
   a=rtpmap:100 t140e/1000
   a=rtpmap:101 red/1000
   a=fmtp:101 103/103/103
   a=fmtp:100 cps=200

   The "text/ulpfec" format includes an indication of how far back the
   redundancy belongs, making it possible to cover bursty packet loss
   better than the other formats with short transmission intervals.  For
   real-time text, it is recommended to send three packets between the
   primary and the redundant transmissions of text.  That makes the
   transmission cover between 500 and 1500 ms of bursty packet loss.
   The variation is because of the varying packet interval between many
   and one simultaneously transmitting source.

   The "text/ulpfec" format has a number of parameters.  One is the
   length of the data to be protected which in this case must be the
   whole t140block.

   Pros:

   The source switching performance is good.  Text from 5 participants
   can be transmitted within 500 ms.

   Good recovery from bursty packet loss.

   The method is based on an existing standard for FEC.

   When more consecutive packet loss than the number of generations of
   redundant data appears, it is possible to deduct the source of the
   lost data when new text arrives from the source.

   Cons:



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   Even if the switching performance is good, it is not as good as for
   the method called "RTP Mixer with multiple primary data in each
   packet" Section 4.1.1.6.  With more than 5 simultaneously sending
   sources, there will be a noticeable delay of text of over 500 ms,
   with 100 ms added per simultaneous source.  This is however beyond
   the requirements and would be a concern only in congestion
   situations.

   The recovery procedure is a bit complex [RFC5109].

   There is more overhead in terms of extra data and extra packets sent
   than in the other methods.  With the recommended two redundant
   generations of data, each packet will be 40 bytes longer than with
   traditional RFC 4103, and at each pause in transmission five extra
   packets with only redundant data will be sent compared to two extra
   packets for the traditional RFC 4103 case.

   A new text media subtype "text/t140e" needs to be registered.

   The processing time in standard organisation will be long.

4.1.1.9.  RTP Mixer indicating participants by a control code in the
          stream

   Text from all participants except the receiving one is transmitted
   from the media mixer in the same RTP session and stream, thus all
   using the same destination address/port combination, the same RTP
   SSRC and , one sequence number series as described in Section 7.1 and
   7.3 of RTP RFC 3550 [RFC3550] about the Mixer function.  The sources
   of the text in each RTP packet are identified by a new defined T.140
   control code "c" followed by a unique identification of the source in
   UTF-8 string format.

   The receiver can use the string for presenting the source of text.
   This method is on the RTP level described in RFC 7667, section 3.6.1
   Media mixing mixer [RFC7667].

   The inline coding of the source of text is applied in the data stream
   itself, and an RTP mixer function is used for coordinating the
   sources of text into one RTP stream.











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   Information uniquely identifying each user in the multi-party session
   is placed as the parameter value "n" in the T.140 application
   protocol function with the function code "c".  The identifier shall
   thus be formatted like this: SOS c n ST, where SOS and ST are coded
   as specified in ITU-T T.140 [T140].  The "c" is the letter "c".  The
   n parameter value is a string uniquely identifying the source.  This
   parameter shall be kept short so that it can be repeated in the
   transmission without concerns for network load.

   A receiving endpoint is supposed to separate text items from the
   different sources and identify and display them accordingly.

   The conference server need to be allowed to decrypt/encrypt the
   packet payload in order to check the source and repack the text.

   Pros:

   If loss of packets occur, it is possible to recover text from
   redundancy at loss of up to the number of redundancy levels carried
   in the RFC 4103 [RFC4103]stream. (normally primary and two redundant
   levels.

   This method can be implemented with most RTP implementations.

   The method can also be used with other transports than RTP

   Cons:

   The method implies a moderate load by the need to insert the source
   often in the stream.

   If more consecutive packet loss than the number of generations of
   redundant data appears, it is not possible to deduct the source of
   the totally lost data.

   The mixer needs to be able to generate suitable and unique source
   identifications which are suitable as labels for the sources.

   Requires an extension on the ITU-T T.140 standard, best made by the
   ITU.

   There is a risk that the control code indicating the change of source
   is lost and the result is false source indication of text.

   The conference server need to be allowed to decrypt/encrypt the
   packet payload.





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4.1.1.10.  Mixing for multi-party unaware user agents

   Multi-party real-time text contents can be transmitted to multi-party
   unaware user agents if source labelling and formatting of the text is
   performed by a mixer.  This method has the limitations that the
   layout of the presentation and the format of source identification is
   purely controlled by the mixer, and that only one source at a time is
   allowed to present in real-time.  Other sources need to be stored
   temporarily waiting for an appropriate moment to switch the source of
   transmitted text.  The mixer controls the switching of sources and
   inserts a source identifier in text format at the beginning of text
   after switch of source.  The logic of the mixer to detect when a
   switch is appropriate should detect a number of places in text where
   a switch can be allowed, including new line, end of sentence, end of
   phrase, a period of inactivity, and a word separator after a long
   time of active transmission.

   This method MAY be used when no support for multi-party awareness is
   detected in the receiving endpoint.The base for his method is
   described in RFC 7667, section 3.6.1 Media mixing mixer [RFC7667].

   See [I-D.ietf-avtcore-multi-party-rtt-mix] for a procedure for mixing
   RTT for a conference-unaware endpoint.

   Pros:

   Can be transmitted to conference-unaware endpoints.

   Can be used with other transports than RTP

   Cons:

   Does not allow full real-time presentation of more than one source at
   a time.  Text from other sources will be delayed.

   The only realistic presentation format is a style with the text from
   the different sources presented with a text label indicating source,
   and the text collected in a chat style presentation but with more
   frequent turn-taking.

   Endpoints often have their own system for adding labels to the RTT
   presentation.  In that case there will be two levels of labels in the
   presentation, one for the mixer and one for the sources.








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   If loss of more packets than can be recovered by the redundancy
   appears, it is not possible to detect which source was struck by the
   loss.  It is also possible that a source switch occurred during the
   loss, and therefore a false indication of the source of text can be
   provided to the user after such loss.

   Because of all these cons, this method is not recommended and should
   be used as the main method, but only as fallback and the last resort
   for backwards interoperability with multi-party unaware endpoints.

   The conference server need to be allowed to decrypt/encrypt the
   packet payload.

4.1.2.  RTP-based bridging with minor RTT media contents reformatting by
        the bridge

   It may be desirable to send text in a multi-party setting in a way
   that allows the text stream contents to be distributed without being
   dealt with in detail in any central server.  A number of such methods
   are described.  However, when writing this specification, no one of
   these methods have a specified way of establishing the session by
   sdp.

4.1.2.1.  RTP Translator sending one RTT stream per participant

   Within the RTP session, text from each participant is transmitted
   from the RTP media translator (bridge) in a separate RTP stream, thus
   using the same destination address/port combination, the same payload
   type number (PT) but separate RTP SSRC parameters and sequence number
   series as described in Section 7.1 and 7.2 of RTP RFC 3550 [RFC3550]
   about the Translator function.  The source of the text in each RTP
   packet is identified by the SSRC parameter in the RTP packets,
   containing the SSRC of the initial source of text.

   A receiving and presenting endpoint is supposed to separate text
   items from the different sources and identify and display them in a
   suitable way.

   This method is described in RFC 7667, section 3.5.1 Relay-transport
   translator or 3.5.2 Media translator [RFC7667].

   The identification of the source is made through the SSRC.  The
   translation to a readable label can be done by mapping to information
   from the RTCP SDES CNAME and NAME packets as described in
   RTP[RFC3550], and also through information in the text media member
   in the conference notification described in RFC 4575 [RFC4575].





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   The sdp exchange for establishing this mixing type can be equal to
   what is used for basic two-party use of RFC 4103 with just an added
   attribute for indicating multi-party capability.

   m=text 49170 RTP/AVP 98 103
   a=rtpmap:98 red/1000
   a=fmtp:98 103/103/103
   a=rtpmap:103 t140/1000
   a=fmtp:103 cps=150
   a=RTT-mix:RTP-translator

   A similar answer including the same RTT-mix attribute would indicate
   that multi-party coding can begin.  An answer without the same RTT-
   mix attribute could result in diversion to use of the mixing method
   for multi-party unaware endpoints Section 4.1.1.10 if more than two
   parties are involved in the session.

   The bridge can add new sources in the communication to a participant
   by first sending a conference notification according to RFC 4575
   [RFC4575] with the SSRC of the new source included in the
   corresponding "text" media member, or by sending an RTCP message with
   the new SSRC in an SDES packet.

   A receiver should be prepared to receive such indications of new
   streams being added to the multi-party session, so that the new SSRC
   is not taken for a change in SSRC value for an already established
   RTP stream.

   Transmission, reception, packet loss recovery and text loss
   indication is performed per source in the separate RTP streams in the
   same way as in two-party sessions with RFC 4103 [RFC4575].

   Text is recommended to be sent by the bridge as soon as it is
   available for transmission, but not less than 250 ms after a previous
   transmission.  This will in many cases result in close to 0 added
   delay by the bridge, because most RTT senders use a 300 ms
   transmission interval.

   It is sometimes said that this configuration is not supported by
   current media declarations in sdp.  RFC 3264 [RFC3264]specifies in
   some places that one media description is supposed to describe just
   one RTP media stream.  However this is not directly referencing an
   RTP stream, and use of multiple RTP streams in the same RTP session
   is recommended in many other RFCs.

   This confusion is clarified in RFC 5576 [RFC5576] section 3 by the
   following statements:




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   "The term "media stream" does not appear in the SDP specification
   itself, but is used by a number of SDP extensions, for instance,
   Interactive Connectivity Establishment (ICE) [ICE], to denote the
   object described by an SDP media description.  This term is
   unfortunately rather confusing, as the RTP specification [RFC3550]
   uses the term "media stream" to refer to an individual media source
   or RTP packet stream, identified by an SSRC, whereas an SDP media
   stream describes an entire RTP session, which can contain any number
   of RTP sources."

   In most cases, it will be sufficient that new sources are introduced
   with a conference notification or RTCP message.  However, RFC 5576
   [RFC5576] specifies attributes which may be used to more explicitly
   announce new sources or restart of earlier established RTP streams.

   This method is encouraged by draft-ietf-avtcore-multiplex-guidelines
   [I-D.ietf-avtcore-multiplex-guidelines] section 5.2.

   Normal operation will be that the bridge receives text packets from
   the source and handles any text recovery and indication of loss
   needed before queueing the resulting clean text for transmission from
   the bridge to the receivers.

   It may however also be possible for the bridge to just convey the
   packet contents as received from the sources, with minor adjustments,
   and let the receiving endpoint handle all aspects of recovery and
   indication of loss, even for the source to bridge path.  In that case
   also the sequence number must be maintained as it was at reception in
   the bridge.  This mode needs further study before application.

   Pros:

   This method is the natural way to do multi-party bridging with RFC
   4103 based RTT.  Only a small addition is included in the session
   establishment to verify capability by the parties because many
   implementations are done without multi-party capability.

   This method has moderate overhead in terms of work for the mixer, but
   high in terms of packet transmission rate.  Five sources sending
   simultaneously cause the bridge to send 15 packets per second to each
   receiver.

   When loss of packets occur, it is possible to recover text from
   redundancy at loss of up to the number of redundancy levels carried
   in the RFC 4103 [RFC4103] stream(normally primary and two redundant
   levels).





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   More loss than what can be recovered, can be detected and the marker
   for text loss can be inserted in the correct stream.

   It may be possible in some scenarios to keep the text encrypted
   through the Translator.

   Minimal delay.  The delay can often be kept close to 0 with at least
   5 simultaneous sending participants.

   Cons:

   There are RTP implementations not supporting the Translator model.
   They will need to use the fall-back to multi-party-unaware mixing.
   An investigation about how common this is is needed before the method
   is used.

   The processing time in standard organisation will be long.

   With many simultaneous sending sources, the total rate of packets
   will be high, and can cause congestion.  The requirement to handle 3
   simultaneous sources in this specification will cause 10 packets per
   second that is manageable in most cases, e.g. considering that audio
   usually use 50 packets per second.

4.1.2.2.  Distributing packets in an end-to-end encryption structure

   In order to achieve end-to-end encryption, it is possible to let the
   packets from the sources just pass though a central distributor, and
   handle the security agreements between the participants.
   Specifications exist for a framework with this functionality for
   application on RTP based conferences in
   [I-D.ietf-perc-private-media-framework].  The RTP flow and mixing
   characteristics has similarities with the method described under "RTP
   Translator sending one RTT stream per participant" above.  RFC 4103
   RTP streams [RFC4103] would fit into the structure and it would
   provide a base for end-to-end encrypted rtt multi-party conferencing.

   Pros:

   Good security

   Straightforward multi-party handling.

   Cons:

   Does not operate under the usual SIP central conferencing
   architecture.




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   Requires the participants to perform a lot of key handling.

   Is work in progress when this is written.

4.1.2.3.  Mesh of RTP endpoints

   Text from all participants are transmitted directly to all others in
   one RTP session, without a central bridge.  The sources of the text
   in each RTP packet are identified by the source network address and
   the SSRC.

   This method is described in RFC 7667, section 3.4 Point to multi-
   point using mesh [RFC7667].

   Pros:

   When loss of packets occur, it is possible to recover text from
   redundancy at loss of up to the number of redundancy levels carried
   in the RFC 4103 [RFC4103] stream. (normally primary and two redundant
   levels.

   This method can be implemented with most RTP implementations.

   Transmitted text can also be used with other transports than RTP

   Cons:

   This model is not described in IMS, NENA and EENA specifications, and
   does therefore not meet the requirements.

   Requires a drastically increasing number of connections when the
   number of participants increase.

4.1.2.4.  Multiple RTP sessions, one for each participant

   Text from all participants are transmitted directly to all others in
   one RTP session each, without a central bridge.  Each session is
   established with a separate media description in SDP.  The sources of
   the text in each RTP packet are identified by the source network
   address and the SSRC.

   Pros:

   When loss of packets occur, it is possible to recover text from
   redundancy at loss of up to the number of redundancy levels carried
   in the RFC 4103 [RFC4103] stream. (normally primary and two redundant
   levels.




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   Complete loss of text can be indicated in the received stream.

   This method can be implemented with most RTP implementations.

   End-to-end encryption is achievable.

   Cons:

   This method is not described in IMS, NENA and ETSI specifications and
   does therefore not meet the requirements.

   A lot of network resources are spent on setting up separate sessions
   for each participant.

5.  Preferred RTP-based multi-party RTT transport method

   For RTP transport of RTT using RTP-mixer technology, one method for
   multi-party mixing and transport stand out as fulfilling the goals
   best and is therefore recommended.  That is: "RTP Mixer using the
   default method but decreased transmission interval" Section 4.1.1.2

   For RTP transport in separate streams or sessions, no current
   recommendation can be made.  A bridging method in the process of
   standardisation with interesting characteristics is the end-to-end
   encryption model "perc" Section 4.1.2.2.

6.  Session control of RTP-based multi-party RTT sessions

   General session control aspects for multi-party sessions are
   described in RFC 4575 [RFC4575] A Session Initiation Protocol (SIP)
   Event Package for Conference State, and RFC 4579 [RFC4579] Session
   Initiation Protocol (SIP) Call Control - Conferencing for User
   Agents.  The nomenclature of these specifications are used here.

   The procedures for a multi-party aware model for RTT-transmission
   shall only be applied if a capability exchange for multi-party aware
   real-time text transmission has been completed and a supported method
   for multi-party real-time text transmission can be negotiated.













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   A method for detection of conference-awareness for centralized SIP
   conferencing in general is specified in RFC 4579 [RFC4579].  The
   focus sends the "isfocus" feature tag in a SIP Contact header.  This
   causes the conference-aware endpoint to subscribe to conference
   notifications from the focus.  The focus then sends notifications to
   the endpoint about entering and disappearing conference participants
   and their media capabilities.  The information is carried XML-
   formatted in a 'conference-info' block in the notification according
   to RFC 4575 [RFC4575].  The mechanism is described in detail in RFC
   4575 [RFC4575].

   Before a conference media server starts sending multi-party RTT to an
   endpoint, a verification of its ability to handle multi-party RTT
   must be made.  A decision on which mechanism to use for identifying
   text from the different participants must also be taken, implicitly
   or explicitly.  These verifications and decisions can be done in a
   number of ways.  The most apparent ways are specified here and their
   pros and cons described.  One of the methods is selected to be the
   one to be used by implementations of the centralized conference model
   according to this specification.

6.1.  Implicit RTT multi-party capability indication

   Capability for RTT multi-party handling can be decided to be
   implicitly indicated by session control items.

   The focus may implicitly indicate muti-party RTT capability by
   including the media child with value "text" in the RFC 4575 [RFC4575]
   conference-info provided in conference notifications.

   An endpoint may implicitly indicate multi-party RTT capability by
   including the text media in the SDP in the session control
   transactions with the conference focus after the subscription to the
   conference has taken place.

   The implicit RTT capability indication means for the focus that it
   can handle multi-party RTT according to the preferred method
   indicated in the RTT multi-party methods section above.

   The implicit RTT capability indication means for the endpoint that it
   can handle multi-party RTT according to the preferred method
   indicated in the RTT multi-party methods section above.

   If the focus detects that an endpoint implicitly declared RTT multi-
   party capability, it SHALL provide RTT according to the preferred
   method.





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   If the focus detects that the endpoint does not indicate any RTT
   multi-party capability, then it shall either provide RTT multi-party
   text in the way specified for conference-unaware endpoint above, or
   refuse to set up the session.

   If the endpoint detects that the focus has implicitly declared RTT
   multi-party capability, it shall be prepared to present RTT in a
   multi-party fashion according to the preferred method.

   Pros:

   Acceptance of implicit multi-party capability implies that no
   standardisation of explicit RTT multi-party capability exchange is
   required.

   Cons:

   If other methods for multi-party RTT are to be used in the same
   implementation environment as the preferred ones, then capability
   exchange needs to be defined for them.

   Cannot be used outside a strictly applied SIP central conference
   model.

6.2.  RTT multi-party capability declared by SIP media-tags

   Specifications for RTT multi-party capability declarations can be
   agreed for use as SIP media feature tags, to be exchanged during SIP
   call control operation according to the mechanisms in RFC 3840
   [RFC3840] and RFC 3841 [RFC3841].  Capability for the RTT Multi-party
   capability is then indicated by the media feature tag "rtt-mix", with
   a set of possible values for the different possible methods.

   The possible values in the list may for example be:

      rtp-mixer

      perc

   rtp-mixer indicates capability for using the RTP-mixer based
   presentation of multi-party text.

   perc indicates capability for using the perc based transmission of
   multi-party text.

   Example: Contact: <sip:a2@beco.example.com>

   ;methods="INVITE,ACK,OPTIONS,BYE,CANCEL"



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   ;+sip.rtt-mix="rtp-mixer"

   If, after evaluation of the alternatives in this specification, only
   one mixing method is selected to be brought to implementation, then
   the media tag can be reduced to a single tag with no list of values.


   An offer-answer exchange should take place and the common method
   selected by the answering party shall be used in the session with
   that UA.

   When no common method is declared, then only the fallback method for
   multi-party unaware participants can be used, or the session dropped.

   If more than one text media section is included in SDP, all must be
   capable of using the declared RTT multi-party method.

   Pros:

   Provides a clear decision method.

   Can be extended with new mixing methods.

   Can guide call routing to a suitable capable focus.

   Cons:

   Requires standardization and IANA registration.

   Is not stream specific.  If more than one text stream is specified,
   all must have the same type of multi-party capability.

   Cannot be used in the WebRTC environment.

6.3.  SDP media attribute for RTT multi-party capability indication

   An attribute can be specified on media level, to be used in text
   media SDP declarations for negotiating RTT multi-party capabilities.
   The attribute can have the name "rtt-mix".

   More than one attribute can be included in one media description.

   The attribute can have a value.  The value can for example be:

      rtp-mixer

      rtp-translator




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      perc

   rtp-mixer indicates capability for using the RTP-mixer and CSRC-list
   based mixing of multi-party text.

   rtp-translator indicates capability for using the RTP-translator
   based mixing

   perc indicates capability for using the perc based transmission of
   multi-party text.


   An offer-answer exchange should take place and the common method
   selected by the answering party shall be used in the session with
   that endpoint.

   When no common method is declared, then only the fallback method for
   multi-party unaware endpoints can be used.

   Example: a=rtt-mix:rtp-mixer

   If, after evaluation of the alternatives in this specification, only
   one mixing method is selected to be brought to implementation, then
   the attribute can be reduced to a single attribute with no list of
   values.

   Pros:

   Provides a clear decision method.

   Can be extended with new mixing methods.

   Can be used on specific text media.

   Can be used also for SDP-controlled WebRTC sessions with multiple
   streams in the same data channel.

   Cons:

   Requires standardization and IANA registration.

   Cannot guide SIP routing.









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6.4.  Simplified SDP media attribute for RTT multi-party capability
      indication

   An attribute can be specified on media level, to be used in text
   media SDP declarations for negotiating RTT multi-party capabilities.
   The attribute can have a name suitable for the selected method and no
   value.  It would be selected and used if only one method for multi-
   party rtt is brought forward from this specification, and the other
   suppressed or found to be possible to negotiate in another way.

   An offer-answer exchange should take place and if both parties
   specify rtt-mixing capability with the same attribute, the selected
   mixing method shall be used.

   When no common method is declared, then only the fallback method for
   multi-party unaware endpoints can be used, or the session not
   accepted for multi-party use.

   Example: a=rtt-mix-rtp-mixer

   Pros:

   Provides a clear decision method.

   Very simple syntax and semantics.

   Can be used on specific text media.

   Cons:

   Requires standardization and IANA registration.

   If another RTT mixing method is also specified in the future, then
   that method may also need to specify and register its own attribute,
   instead of if an attribute with a parameter value is used, when only
   an addition of a new possible value is needed.

   Cannot guide SIP routing.

6.5.  SDP format parameter for RTT multi-party capability indication

   An FMTP format parameter can be specified for the RFC 4103
   [RFC4103]media, to be used in text media SDP declarations for
   negotiating RTT multi-party capabilities.  The parameter can have the
   name "rtt-mix", with one or more of its possible values.

   The possible values in the list are:




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      rtp-mixer

      perc

   rtp-mixer indicates capability for using the RTP-mixer based mixing
   and presentation of multi-party text using the CSRC-list.

   perc indicates capability for using the perc based transmission of
   multi-party text.

   Example: a=fmtp 96 98/98/98 rtt-mix=rtp-mixer

   If, after evaluation of the alternatives in this specification, only
   one mixing method is selected to be brought to implementation, then
   the parameter can be reduced to a single parameter with no list of
   values.


   An offer-answer exchange should take place and the common method
   selected by the answering party shall be used in the session with
   that UA.

   When no common method is declared, then only the fallback method can
   be used, or the session denied.

   Pros:

   Provides a clear decision method.

   Can be extended with new mixing methods.

   Can be used on specific text media.

   Can be used also for SDP-controlled WebRTC sessions with multiple
   streams in the same data channel.

   Cons:

   Requires standardization and IANA registration.

   May cause interop problems with current RFC4103 [RFC4103]
   implementations not expecting a new fmtp-parameter.

   Cannot guide SIP routing.







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6.6.  A text media subtype for support of multi-party rtt

   Indicating a specific text media subtype in SDP is a straightforward
   way for negotiating multi-party capability.  Especially if there are
   format differences from the "text/red" and "text/t140" formats of
   RFC4103 [RFC4103], then this is a natural way to do the negotiation
   for multi-party rtt.


   Pros:

   No extra efforts if a new format is needed anyway.

   Cons:

   None specific to using the format indication for negotiation of
   multi-party capability.  But only feasible if a new format is needed
   anyway.

6.7.  Preferred capability declaration method for RTP-based transport.

   If the preferred transport method is one with a specific media
   subtype in sdp, then speciication by media subtype is preferred.

   If this would not be the case, then the preferred capability
   declaration method would be the one with a specific SDP attribute for
   the selected mixing method Section 6.4 because it is straightforward.

6.8.  Identification of the source of text for RTP-based solutions

   The main way to identify the source of text in the RTP based solution
   is by the SSRC of the sending participant.  In the RTP-mixer
   solution, this SSRC is included in the CSRC list of the transmitted
   packets.  Further identification that may be needed for better
   labelling of received text may be achieved from a number of sources.
   It may be the RTCP SDES CNAME and NAME reports, and in the conference
   notification data (RFC 4575) [RFC4575].

   As soon as a new member is added to the RTP session, its
   characteristics should be transmitted in RTCP SDES CNAME and NAME
   reports according to section 6.5 in RFC 3550 [RFC3550].  The
   information about the participant should also be included in the
   conference data including the text media member in a notification
   according to RFC 4575 [RFC4575].







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   The RTCP SDES report, SHOULD contain identification of the source
   represented by the SSRC/CSRC identifier.  This identification MUST
   contain the CNAME field and MAY contain the NAME field and other
   defined fields of the SDES report.

   A focus UA SHOULD primarily convey SDES information received from the
   sources of the session members.  When such information is not
   available, the focus UA SHOULD compose SSRC/CSRC, CNAME and NAME
   information from available information from the SIP session with the
   participant.

   Provision of detailed information in the NAME field has security
   implications, especially if provided without encryption.

7.  RTT bridging in WebRTC

   Within WebRTC, real-time text is specified to be carried in WebRTC
   data channels as specified in
   [I-D.ietf-mmusic-t140-usage-data-channel].  A few ways to handle
   multi-party RTT are mentioned briefly.  They are repeated below.

7.1.  RTT bridging in WebRTC with one data channel per source

   A straightforward way to handle multi-party RTT is for the bridge to
   open one T.140 data channel per source towards the receiving
   participants.

   The stream-id forms a unique stream identification.

   The identification of the source is made through the Label property
   of the channel, and session information belonging to the source.  The
   endpoint can compose a readable label for the presentation from this
   information.

   Pros:

   This is a straightforward solution.

   The load per source is low.

   Cons:

   With a high number of participants, the overhead of establishing and
   maintaining the high number of data channels required may be high,
   even if the load per channel is low.






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7.2.  RTT bridging in WebRTC with one common data channel

   A way to handle multi-party RTT in WebRTC is for the bridge combine
   text from all sources into one data channel and insert the sources in
   the stream by a T.140 control code for source.

   This method is described in a corresponding section for RTP
   transmission above in Section 4.1.1.9.

   The identification of the source is made through insertion in the
   beginning of each text transmission from a source of a control code
   extension "c" followed by a string representing the source, framed by
   the control code start and end flags SOS and ST (See ITU-T T.140
   [T140]).

   A receiving endpoint is supposed to separate text items from the
   different sources and identify and display them in a suitable way.

   The endpoint does not always display the source identification in the
   received text at the place where it is received, but has the
   information as a guide for planning the presentation of received
   text.  A label corresponding to the source identification is
   presented when needed depending on the selected presentation style.

   Pros:

   This solution has relatively low overhead on session and network
   level

   Cons:

   This solution has higher overhead on the media contents level than
   the WebRTC solution above.

   Standardisation of the new control code "c" in ITU-T T.140 [T140] is
   required.

   The conference server need to be allowed to decrypt/encrypt the data
   channel contents.

7.3.  Preferred rtt multi-party method for WebRTC

   For WebRTC, one method is to prefer because of the simplicity.  So,
   for WebRTC, the method to implement for multi-party RTT with multi-
   party aware parties when no other method is explicitly agreed between
   implementing parties is: "RTT bridging in WebRTC with one data
   channel per source" Section 7.1.




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8.  Presentation of multi-party text

   All session participants with RTP based transport MUST observe the
   SSRC/CSRC field of incoming text RTP packets, and make note of which
   source they came from in order to be able to present text in a way
   that makes it easy to read text from each participant in a session,
   and get information about the source of the text.

   In the WebRTC case, the Label parameter and other provided endpoint
   information should be used for the same purpose.

8.1.  Associating identities with text streams

   A source identity SHOULD be composed from available information
   sources and displayed together with the text as indicated in ITU-T
   T.140 Appendix[T140].

   The source identity should primarily be the NAME field from incoming
   SDES packets.  If this information is not available, and the session
   is a two-party session, then the T.140 source identity SHOULD be
   composed from the SIP session participant information.  For multi-
   party sessions the source identity may be composed by local
   information if sufficient information is not available in the
   session.

   Applications may abbreviate the presented source identity to a
   suitable form for the available display.

   Applications may also replace received source information with
   internally used nicknames.

8.2.  Presentation details for multi-party aware endpoints.

   The multi-party aware endpoint should after any action for recovery
   of data from lost packets, separate the incoming streams and present
   them according to the style that the receiving application supports
   and the user has selected.  The decisions taken for presentation of
   the multi-party interchange shall be purely on the receiving side.
   The sending application must not insert any item in the stream to
   influence presentation that is not requested by the sending
   participant.










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8.2.1.  Bubble style presentation

   One often used style is to present real-time text in chunks in
   readable bubbles identified by labels containing names of sources.
   Bubbles are placed in one column in the presentation area and are
   closed and moved upwards in the presentation area after certain items
   or events, when there is also newer text from another source that
   would go into a new bubble.  The text items that allows bubble
   closing are any character closing a phrase or sentence followed by a
   space or a timeout of a suitable time (about 10 seconds).

   Real-time active text sent from the local user should be presented in
   a separate area.  When there is a reason to close a bubble from the
   local user, the bubble should be placed above all real-time active
   bubbles, so that the time order that real-time text entries were
   completed is visible.

   Scrolling is usually provided for viewing of recent or older text.
   When scrolling is done to an earlier point in the text, the
   presentation shall not move the scroll position by new received text.
   It must be the decision of the local user to return to automatic
   viewing of latest text actions.  It may be useful with an indication
   that there is new text to read after scrolling to an earlier position
   has been activated.

   The presentation area may become too small to present all text in all
   real-time active bubbles.  Various techniques can be applied to
   provide a good overview and good reading opportunity even in such
   situations.  The active real-time bubble may have a limited number of
   lines and if their contents need more lines, then a scrolling
   opportunity within the real-time active bubble is provided.  Another
   method can be to only show the label and the last line of the active
   real-time bubble contents, and make it possible to expand or compress
   the bubble presentation between full view and one line view.

   Erasures require special consideration.  Erasure within a real-time
   active bubble is straightforward.  But if erasure from one
   participant affects the last character before a bubble, the whole
   previous bubble becomes the actual bubble for real-time action by
   that participant and is placed below all other bubbles in the
   presentation area.  If the border between bubbles was caused by the
   CRLF characters (instead of the normal "Line Separator"), only one
   erasure action is required to erase this bubble border.  When a
   bubble is closed, it is moved up, above all real-time active bubbles.

   A three-party view is shown in this example .





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                 _________________________________________________
                |                                              |^|
                |                                              |-|
                |[Alice] Hi, Alice here.                       | |
                |                                              | |
                |[Bob] Bob as well.                            | |
                |                                              | |
                |[Eve] Hi, this is Eve, calling from Paris.    | |
                |      I thought you should be here.           | |
                |                                              | |
                |[Alice] I am coming on Thursday, my           | |
                |      performance is not until Friday morning.| |
                |                                              | |
                |[Bob] And I on Wednesday evening.             | |
                |                                              | |
                |[Alice] Can we meet on Thursday evening?      | |
                |                                              | |
                |[Eve] Yes, definitely. How about 7pm.         | |
                |     at the entrance of the restaurant        | |
                |     Le Lion Blanc?                           | |
                |[Eve] we can have dinner and then take a walk | |
                |                                              | |
                | <Eve-typing> But I need to be back to        | |
                |    the hotel by 11 because I need            | |
                |                                              | |
                | <Bob-typing> I wou                           |-|
                |______________________________________________|v|
                | of course, I underst                           |
                |________________________________________________|

               Figure 1: Three-party call with bubble style.

   Figure 1: Example of a three-party call presented in the bubble
   style.

8.2.2.  Other presentation styles

   Other presentation styles than the bubble style may be arranged and
   appreciated by the users.  In a video conference one way may be to
   have a real-time text area below the video view of each participant.
   Another view may be to provide one column in a presentation area for
   each participant and place the text entries in a relative vertical
   position corresponding to when text entry in them was completed.  The
   labels can then be placed in the column header.  The considerations
   for ending and moving and erasure of entered text discussed above for
   the bubble style are valid also for these styles.

   This figure shows how a coordinated column view MAY be presented.



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   _____________________________________________________________________
   |       Bob          |       Eve            |       Alice           |
   |____________________|______________________|_______________________|
   |                    |                      |I will arrive by TGV.  |
   |My flight is to Orly|                      |Convenient to the main |
   |                    |Hi all, can we plan   |station.               |
   |                    |for the seminar?      |                       |
   |Eve, will you do    |                      |                       |
   |your presentation on|                      |                       |
   |Friday?             |Yes, Friday at 10.    |                       |
   |Fine, wo            |                      |We need to meet befo   |
   |___________________________________________________________________|

   Figure 2: A coordinated column-view of a three-party session with
   entries ordered in approximate time-order.

9.  Presentation details for multi-party unaware endpoints.

   Multi-party unaware endpoints are prepared only for presentation of
   two sources of text, the local user and a remote user.  If mixing for
   multi-party unaware endpoints is to be supported, in order to enable
   some multi-party communication with such endpoint, the mixer need to
   plan the presentation and insert labels and line breaks before
   lables.  Many limitations appear for this presentation mode, and it
   must be seen as a fallback and a last resort.

   A procedure for presenting RTT to a conference-unaware endpoint is
   included in [I-D.ietf-avtcore-multi-party-rtt-mix]

10.  Security Considerations

   The security considerations valid for RFC 4103 [RFC4103] and RFC 3550
   [RFC3550] are valid also for the multi-party sessions with text.

11.  IANA Considerations

   The items for indication and negotiation of capability for multi-
   party rtt should be registered with IANA in the specifications where
   they are specified in detail.












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12.  Congestion considerations

   The congestion considerations described in RFC 4103 [RFC4103] are
   valid also for the recommended RTP-based multi-party use of the real-
   time text transport.  A risk for congestion may appear if a number of
   conference participants are active transmitting text simultaneously,
   because the recommended RTP-based multi-party transmission method
   does not allow multiple sources of text to contribute to the same
   packet.

   In situations of risk for congestion, the Focus UA MAY combine
   packets from the same source to increase the transmission interval
   per source up to one second.  Local conference policy in the Focus UA
   may be used to decide which streams shall be selected for such
   transmission frequency reduction.

13.  Acknowledgements

   Arnoud van Wijk for contributions to an earlier, expired draft of
   this memo.

14.  Change history

14.1.  Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-03

   Modified info on the method with RFC 4103 format and sdp attribute
   "rtt-mix-rtp-mixer".

   Increased the performance requirements section.

   Inserted recommendations, with emphasis on ease of implementation and
   ease of standardisation.

14.2.  Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-02

   Added detail in the section on RTP translator model alternative
   4.1.2.1.

14.3.  Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-01

   Added three more methods for RTP-mixer mixing.  Two RFC 5109 FEC
   based and another with modified data header to detect source of
   completely lost text.

   Separated RTP-based and WebRTC based solutions.






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   Deleted the multi-party-unaware mixing procedure appendix.  It is now
   included in the draft draft-ietf-avtcore-multi-party-rtt-mix.  Kept a
   section with a reference to the new place.

14.4.  Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to draft-
       hellstrom-avtcore-multi-party-rtt-solutions-00

   Add discussion about switching performance, as discussed in avtcore
   on March 13.

   Added that a decrease of transmission interval to 100 ms increases
   switching performance by a factor 3, but still not sufficient.

   Added that the CSRC-list method also uses 100 milliseconds
   transmission interval.

   Added the method with multiple primary text in each packet.

   Added the timestamp-based method for rtp-mixing proposed by James
   Hamlin on March 14.


   Corrected the chat style presentation example picture.  Delete a few
   "[mix]".

14.5.  Changes from version draft-hellstrom-mmusic-multi-party-rtt-01 to
       -02

   Change from a general overview to overview with clear
   recommendations.

   Splits text coordination methods in three groups.

   Recommends rtt-mixer with sources in CSRC-list but referenes to its
   spec for details.

   Shortened Appendix with conference-unaware example.

   Cleaned up preferences.

   Inserted pictures of screen-views.

15.  References

15.1.  Normative References






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   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

15.2.  Informative References

   [EN301549] ETSI, "EN 301 549. Accessibility requirements for ICT
              products and services", November 2019,
              <https://www.etsi.org/deliver/
              etsi_en/301500_301599/301549/03.01.01_60/
              en_301549v030101p.pdf>.

   [I-D.ietf-avtcore-multi-party-rtt-mix]
              Hellstrom, G., "RTP-mixer formatting of multi-party Real-
              time text", Work in Progress, Internet-Draft, draft-ietf-
              avtcore-multi-party-rtt-mix-06, 11 June 2020,
              <https://tools.ietf.org/html/draft-ietf-avtcore-multi-
              party-rtt-mix-06>.

   [I-D.ietf-avtcore-multiplex-guidelines]
              Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
              and R. Even, "Guidelines for using the Multiplexing
              Features of RTP to Support Multiple Media Streams", Work
              in Progress, Internet-Draft, draft-ietf-avtcore-multiplex-
              guidelines-12, 16 June 2020, <https://tools.ietf.org/html/
              draft-ietf-avtcore-multiplex-guidelines-12>.

   [I-D.ietf-mmusic-t140-usage-data-channel]
              Holmberg, C. and G. Hellstrom, "T.140 Real-time Text
              Conversation over WebRTC Data Channels", Work in Progress,
              Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel-
              14, 10 April 2020, <https://tools.ietf.org/html/draft-
              ietf-mmusic-t140-usage-data-channel-14>.

   [I-D.ietf-perc-private-media-framework]
              Jones, P., Benham, D., and C. Groves, "A Solution
              Framework for Private Media in Privacy Enhanced RTP
              Conferencing (PERC)", Work in Progress, Internet-Draft,
              draft-ietf-perc-private-media-framework-12, 5 June 2019,
              <https://tools.ietf.org/html/draft-ietf-perc-private-
              media-framework-12>.

   [NENAi3]   NENA, "NENA-STA-010.2-2016. Detailed Functional and
              Interface Standards for the NENA i3 Solution", October
              2016, <https://www.nena.org/page/i3_Stage3>.





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   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J.C., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              DOI 10.17487/RFC2198, September 1997,
              <https://www.rfc-editor.org/info/rfc2198>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <https://www.rfc-editor.org/info/rfc3261>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <https://www.rfc-editor.org/info/rfc3264>.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <https://www.rfc-editor.org/info/rfc3550>.

   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840,
              DOI 10.17487/RFC3840, August 2004,
              <https://www.rfc-editor.org/info/rfc3840>.

   [RFC3841]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
              Preferences for the Session Initiation Protocol (SIP)",
              RFC 3841, DOI 10.17487/RFC3841, August 2004,
              <https://www.rfc-editor.org/info/rfc3841>.

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
              <https://www.rfc-editor.org/info/rfc4103>.

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353,
              DOI 10.17487/RFC4353, February 2006,
              <https://www.rfc-editor.org/info/rfc4353>.

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
              Session Initiation Protocol (SIP) Event Package for
              Conference State", RFC 4575, DOI 10.17487/RFC4575, August
              2006, <https://www.rfc-editor.org/info/rfc4575>.





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   [RFC4579]  Johnston, A. and O. Levin, "Session Initiation Protocol
              (SIP) Call Control - Conferencing for User Agents",
              BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006,
              <https://www.rfc-editor.org/info/rfc4579>.

   [RFC4597]  Even, R. and N. Ismail, "Conferencing Scenarios",
              RFC 4597, DOI 10.17487/RFC4597, August 2006,
              <https://www.rfc-editor.org/info/rfc4597>.

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <https://www.rfc-editor.org/info/rfc5109>.

   [RFC5194]  van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real-
              Time Text over IP Using the Session Initiation Protocol
              (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008,
              <https://www.rfc-editor.org/info/rfc5194>.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
              <https://www.rfc-editor.org/info/rfc5576>.

   [RFC6443]  Rosen, B., Schulzrinne, H., Polk, J., and A. Newton,
              "Framework for Emergency Calling Using Internet
              Multimedia", RFC 6443, DOI 10.17487/RFC6443, December
              2011, <https://www.rfc-editor.org/info/rfc6443>.

   [RFC6881]  Rosen, B. and J. Polk, "Best Current Practice for
              Communications Services in Support of Emergency Calling",
              BCP 181, RFC 6881, DOI 10.17487/RFC6881, March 2013,
              <https://www.rfc-editor.org/info/rfc6881>.

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,
              <https://www.rfc-editor.org/info/rfc7667>.

   [T140]     ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for
              multimedia application text conversation", February 1998,
              <https://www.itu.int/rec/T-REC-T.140-199802-I/en>.

   [T140ad1]  ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000),
              Protocol for multimedia application text conversation",
              February 2000,
              <https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>.






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   [TS103479] ETSI, "TS 103 479. Emergency communications (EMTEL); Core
              elements for network independent access to emergency
              services", December 2019, <https://www.etsi.org/deliver/
              etsi_ts/103400_103499/103479/01.01.01_60/
              ts_103479v010101p.pdf>.

   [TS22173]  3GPP, "IP Multimedia Core Network Subsystem (IMS)
              Multimedia Telephony Service and supplementary services;
              Stage 1", 3GPP TS 22.173 17.1.0, 20 December 2019,
              <http://www.3gpp.org/ftp/Specs/html-info/22173.htm>.

   [TS24147]  3GPP, "Conferencing using the IP Multimedia (IM) Core
              Network (CN) subsystem; Stage 3", 3GPP TS 24.147 16.0.0,
              19 December 2019,
              <http://www.3gpp.org/ftp/Specs/html-info/24147.htm>.

Author's Address

   Gunnar Hellstrom
   Gunnar Hellstrom Accessible Communication
   Esplanaden 30
   SE-136 70 Vendelso
   Sweden

   Phone: +46 708 204 288
   Email: gunnar.hellstrom@ghaccess.se

























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