AVTCore                                                     G. Hellstrom
Internet-Draft                 Gunnar Hellstrom Accessible Communication
Updates: 4103 (if approved)                                11                                28 April 2021
Intended status: Standards Track
Expires: 13 30 October 2021

           RTP-mixer formatting of multi-party Real-time text


   Enhancements for RFC 4103 real-time text mixing are provided in this
   document, suitable for a centralized conference model that enables
   source identification and rapidly interleaved transmission of text
   from different sources.  The intended use is for real-time text
   mixers and participant endpoints capable of providing an efficient
   presentation or other treatment of a multi-party real-time text
   session.  The specified mechanism builds on the standard use of the
   CSRC list in the RTP packet for source identification.  The method
   makes use of the same "text/t140" and "text/red" formats as for two-
   party sessions.

   Solutions using multiple RTP streams in the same RTP session are
   briefly mentioned, as they could have some benefits over the RTP-
   mixer model.  The possibility to implement the solution in a wide
   range of existing RTP implementations made the RTP-mixer model be
   selected to be fully specified in this document.

   A capability exchange is specified so that it can be verified that a
   mixer and a participant can handle the multi-party coded real-time
   text stream using the RTP-mixer method.  The capability is indicated
   by use of an SDP media attribute "rtt-mixer".

   The document updates RFC 4103 "RTP Payload for Text Conversation".

   A specification of how a mixer can format text for the case when the
   endpoint is not multi-party aware is also provided.

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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   This Internet-Draft will expire on 13 30 October 2021.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   6
     1.2.  Selected solution and considered alternatives . . . . . .   6
     1.3.  Intended application  . . . . . . . . . . . . . . . . . .   9
   2.  Overview of the two specified solutions and selection of
           method  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     2.1.  The RTP-mixer based solution for multi-party aware
           endpoints . . . . . . . . . . . . . . . . . . . . . . . .  10
     2.2.  Mixing for multi-party unaware endpoints  . . . . . . . .  10
     2.3.  Offer/answer considerations . . . . . . . . . . . . . . .  11
     2.4.  Actions depending on capability negotiation result  . . .  11
   3.  Details for the RTP-mixer based multi-party aware mixing
           method  . . . . . . . . . . . . . . . . . . . . . . . . .  12
     3.1.  Use of fields in the RTP packets  . . . . . . . . . . . .  12
     3.2.  Initial transmission of a BOM character . . . . . . . . .  12
     3.3.  Keep-alive  . . . . . . . . . . . . . . . . . . . . . . .  12
     3.4.  Transmission interval . . . . . . . . . . . . . . . . . .  13
     3.5.  Only one source per packet  . . . . . . . . . . . . . . .  13
     3.6.  Do not send received text to the originating source . . .  13
     3.7.  Clean incoming text . . . . . . . . . . . . . . . . . . .  13
     3.8.  Redundant transmission principles . . . . . . . . . . . .  13
     3.9.  Interleaving text from different sources  . . . . . . . .  14
     3.10. Text placement in packets . . . . . . . . . . . . . . . .  14
     3.11. Empty T140blocks  . . . . . . . . . . . . . . . . . . . .  15
     3.12. Creation of the redundancy  . . . . . . . . . . . . . . .  15
     3.13. Timer offset fields . . . . . . . . . . . . . . . . . . .  15
     3.14. Other RTP header fields . . . . . . . . . . . . . . . . .  16
     3.15. Pause in transmission . . . . . . . . . . . . . . . . . .  16
     3.16. RTCP considerations . . . . . . . . . . . . . . . . . . .  16
     3.17. Reception of multi-party contents . . . . . . . . . . . .  16
     3.18. Performance considerations  . . . . . . . . . . . . . . .  18
     3.19. Security for session control and media  . . . . . . . . .  19
     3.20. SDP offer/answer examples . . . . . . . . . . . . . . . .  19
     3.21. Packet sequence example from interleaved transmission . .  20  21
     3.22. Maximum character rate "CPS"  . . . . . . . . . . . . . .  23  24
   4.  Presentation level considerations . . . . . . . . . . . . . .  23  24
     4.1.  Presentation by multi-party aware endpoints . . . . . . .  24  25
     4.2.  Multi-party mixing for multi-party unaware endpoints  . .  26  27
   5.  Relation to Conference Control  . . . . . . . . . . . . . . .  31  33
     5.1.  Use with SIP centralized conferencing framework . . . . .  32  33
     5.2.  Conference control  . . . . . . . . . . . . . . . . . . .  32  33
   6.  Gateway Considerations  . . . . . . . . . . . . . . . . . . .  32  33
     6.1.  Gateway considerations with Textphones (e.g.  TTYs).  . .  32  33
     6.2.  Gateway considerations with WebRTC. . . . . . . . . . . .  32  34
   7.  Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . .  33  35
   8.  Congestion considerations . . . . . . . . . . . . . . . . . .  34  35
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  34  35
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  34  35
     10.1.  Registration of the "rtt-mixer" SDP media attribute  . .  34  35
   11. Security Considerations . . . . . . . . . . . . . . . . . . .  35  36
   12. Change history  . . . . . . . . . . . . . . . . . . . . . . .  35  37
     12.1.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . .  35  37
     12.2.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . .  36  37
     12.3.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . .  36  37
     12.4.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . .  36  38
     12.5.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . .  36  38
     12.6.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . .  37  38
     12.7.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . .  37  38
     12.8.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . .  37  39
     12.9.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . .  37  39
     12.10. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . .  38  39
     12.11. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . .  38  39

     12.12. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . .  38  40
     12.13. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . .  39  40
     12.14. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . .  41
     12.15. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . .  39
     12.15.  41
     12.16. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-03 to
             draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . .  40
     12.16.  41
     12.17. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-02 to
             -03 . . . . . . . . . . . . . . . . . . . . . . . . . .  40
     12.17.  42
     12.18. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-01 to
             -02 . . . . . . . . . . . . . . . . . . . . . . . . . .  40
     12.18.  42
     12.19. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-00 to
             -01 . . . . . . . . . . . . . . . . . . . . . . . . . .  41  43
   13. References  . . . . . . . . . . . . . . . . . . . . . . . . .  41  43
     13.1.  Normative References . . . . . . . . . . . . . . . . . .  41  43
     13.2.  Informative References . . . . . . . . . . . . . . . . .  43  44
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  43  45

1.  Introduction

   "RTP Payload for Text Conversation" [RFC4103] specifies use of RTP
   [RFC3550] for transmission of real-time text (RTT) and the "text/
   t140" format.  It also specifies a redundancy format "text/red" for
   increased robustness.  The "text/red" format is registered in

   Real-time text is usually provided together with audio and sometimes
   with video in conversational sessions.

   A requirement related to multi-party sessions from the presentation
   level standard T.140 [T140] for real-time text is: "The display of
   text from the members of the conversation should 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

   Another requirement is that the mixing procedure must not introduce
   delays in the text streams that are experienced to be disturbing the
   real-time experience of the receiving users.

   Use of RTT is increasing, and specifically, use in emergency calls is
   increasing.  Emergency call use requires multi-party mixing.  RFC
   4103 "RTP Payload for Text Conversation" mixer implementations can
   use traditional RTP functions for source identification, but the
   performance of the mixer when giving turns for the different sources
   to transmit is limited when using the default transmission
   characteristics with redundancy.

   The redundancy scheme of [RFC4103] enables efficient transmission of
   earlier transmitted redundant text in packets together with new text.
   However the redundancy header format has no source indicators for the
   redundant transmissions.  The redundant parts in a packet must
   therefore be from the same source as the new text.  The recommended
   transmission is one new and two redundant generations of text
   (T140blocks) in each packet and the recommended transmission interval
   for two-party use is 300 ms.

   Real-time text mixers for multi-party sessions need to include the
   source with each transmitted group of text from a conference
   participant so that the text can be transmitted interleaved with text
   groups from different sources in the rate they are created.  This
   enables the text groups to be presented by endpoints in suitable
   grouping with other text from the same source.

   The presentation can then be arranged so that text from different
   sources can be presented in real-time and easily read.  At the same
   time it is possible for a reading user to perceive approximately when
   the text was created in real time by the different parties.  The
   transmission and mixing is intended to be done in a general way so
   that presentation can be arranged in a layout decided by the

   There are existing implementations of RFC 4103 in endpoints without
   the updates from this document.  These will not be able to receive
   and present real-time text mixed for multi-party aware endpoints.

   A negotiation mechanism is therefore needed for verification if the
   parties are able to handle a common method for multi-party
   transmission and agreeing on using that method.

   A fall-back mixing procedure is also needed for cases when the
   negotiation result indicates that a receiving endpoint is not capable
   of handling the mixed format.  Multi-party unaware endpoints would
   possibly otherwise present all received multi-party mixed text as if
   it came from the same source regardless of any accompanying source
   indication coded in fields in the packet.  Or they may have any other
   undesirable way of acting on the multi-party content.  The fall-back
   method is called the mixing procedure for multi-party unaware
   endpoints.  The fall-back method is naturally not expected to meet
   all performance requirements placed on the mixing procedure for
   multi-party aware endpoints.

   The document updates [RFC4103] by introducing an attribute for
   indicating capability for the RTP-mixer based multi-party mixing case
   and rules for source indications and interleaving of text from
   different sources.

1.1.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "OPTIONAL" in this document are to be interpreted as described in BCP
   14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown above.

   mixer, RTP-translator as are defined in [RFC3550] [RFC3550].

   The term "T140block" is defined in [RFC4103] to contain one or more
   T.140 code elements.

   "TTY" stands for a text telephone type used in North America.

   "WebRTC" stands for web based communication specified by W3C and
   IETF.  See [RFC8825].

   "DTLS-SRTP" stands for security specified in [RFC5764].

   "multi-party aware" stands for an endpoint receiving real-time text
   from multiple sources through a common conference mixer being able to
   present the text in real-time separated by source and presented so
   that a user can get an impression of the approximate relative timing
   of text from different parties.

   "multi-party unaware" stands for an endpoint not itself being able to
   separate text from different sources when received through a common
   conference mixer.

1.2.  Selected solution and considered alternatives

   A number of alternatives were considered when searching an efficient
   and easily implemented multi-party method for real-time text.  This
   section explains a few of them briefly.

   Multiple RTP streams, one per participant.
      One RTP stream per source would be sent in the same RTP session
      with the "text/red" format.  From some points of view, use of
      multiple RTP streams, one for each source, sent in the same RTP
      session would be efficient, and would use exactly the same packet
      format as [RFC4103] and the same payload type.  A couple of
      relevant scenarios using multiple RTP-streams are specified in
      "RTP Topologies" [RFC7667].  One possibility of special interest
      is the Selective Forwarding Middlebox (SFM) topology specified in
      RFC 7667 section 3.7 that could enable end to end encryption.  In
      contrast to audio and video, real-time text is only transmitted
      when the users actually transmit information.  Thus an SFM
      solution would not need to exclude any party from transmission
      under normal conditions.  In order to allow the mixer to convey
      the packets with the payload preserved and encrypted, an SFM
      solution would need to act on some specific characteristics of the
      "text/red" format.  The redundancy headers are part of the
      payload, so the receiver would need to just assume that the
      payload type number in the redundancy header is for "text/t140".
      The characters per second parameter (CPS) would need to act per
      stream.  The relation between the SSRC and the source would need
      to be conveyed in some specified way, e.g. in the CSRC.  Recovery
      and loss detection would preferably be based on sequence number
      gap detection.  Thus sequence number gaps in the incoming stream
      to the mixer would need to be reflected in the stream to the
      participant and no new gaps created by the mixer.  However, the
      RTP implementation in both mixers and endpoints need to support
      multiple streams in the same RTP session in order to use this
      mechanism.  For best deployment opportunity, it should be possible
      to upgrade existing endpoint solutions to be multi-party aware
      with a reasonable effort.  There is currently a lack of support
      for multi-stream RTP in certain implementation technologies.  This
      fact made this solution only briefly mentioned in this document as
      an option for further study.

   RTP-mixer based method for multi-party aware endpoints.
      The "text/red" format in RFC 4103 is sent with shorter
      transmission interval with the RTP-mixer method and indicating
      source in CSRC.  The "text/red" format with "text/t140" payload in
      a single RTP stream can be sent when text is available from the
      call participants instead of at the regular 300 ms.  The source is
      indicated in the CSRC field.  Transmission of packets with text
      from different sources can then be done smoothly while
      simultaneous transmission occurs as long as it is not limited by
      the maximum character rate "CPS".  With ten participants sending
      text simultaneously, the switching and transmission performance is
      good.  With more simultaneously sending participants, and
      receivers with default capacity there will be a noticeable
      jerkiness and delay in text presentation.  The jerkiness will be
      more expressed the more participants who send text simultaneously.
      Two seconds jerkiness will be noticeable and slightly unpleasant,
      but corresponds in time to what typing humans often cause by
      hesitation or changing position while typing.  A benefit of this
      method is that no new packet format needs to be introduced and
      implemented.  Since simultaneous typing by more than two parties
      is very rare, this method can be used successfully with good
      performance.  Recovery of text in case of packet loss is based on
      analysis of timestamps of received redundancy versus earlier
      received text.  Negotiation is based on a new SDP media attribute
      "rtt-mixer".  This method is selected to be the main one specified
      in this document.

   Multiple sources per packet.
      A new "text" media subtype would be specified with up to 15
      sources in each packet.  The mechanism would make use of the RTP
      mixer model specified in RTP [RFC3550].  Text from up to 15
      sources can be included in each packet.  Packets are normally sent
      every 300 ms.  The mean delay will be 150 ms.  The sources are
      indicated in strict order in the CSRC list of the RTP packets.  A
      new redundancy packet format is specified.  This method would
      result in good performance, but would require standardisation and
      implementation of new releases in the target technologies that
      would take more time than desirable to complete.  It was therefore
      not selected to be included in this document.

   Mixing for multi-party unaware endpoints
      Presentation of text from multiple parties is prepared by the
      mixer in one single stream.  It is desirable to have a method that
      does not require any modifications in existing user devices
      implementing RFC 4103 for RTT without explicit support of multi-
      party sessions.  This is possible by having the mixer insert a new
      line and a text formatted source label before each switch of text
      source in the stream.  Switch of source can only be done in places
      in the text where it does not disturb the perception of the
      contents.  Text from only one source can be presented in real time
      at a time.  The delay will therefore be varying.  The method also
      has other limitations, but is included in this document as a
      fallback method.  In calls where parties take turns properly by
      ending their entries with a new line, the limitations will have
      limited influence on the user experience. while only two parties
      send text, these two will see the text in real time with no delay.
      This method is specified as a fallback method in this document.

   RTT transport in WebRTC
      Transport of real-time text in the WebRTC technology is specified
      to use the WebRTC data channel in [RFC8865].  That specification
      contains a section briefly describing its use in multi-party
      sessions.  The focus of this document is RTP transport.
      Therefore, even if the WebRTC transport provides good multi-party
      performance, it is just mentioned in this document in relation to
      providing gateways with multi-party capabilities between RTP and
      WebRTC technologies.

1.3.  Intended application

   The method for multi-party real-time text specified in this document
   is primarily intended for use in transmission between mixers and
   endpoints in centralised mixing configurations.  It is also
   applicable between mixers.  An often mentioned application is for
   emergency service calls with real-time text and voice, where a
   calltaker wants to make an attended handover of a call to another
   agent, and stay observing the session.  Multimedia conference
   sessions with support for participants to contribute in text is
   another application.  Conferences with central support for speech-to-
   text conversion is yet another mentioned application.

   In all these applications, normally only one participant at a time
   will send long text utterances.  In some cases, one other participant
   will occasionally contribute with a longer comment simultaneously.
   That may also happen in some rare cases when text is interpreted to
   text in another language in a conference.  Apart from these cases,
   other participants are only expected to contribute with very brief
   utterings while others are sending text.

   Users expect that the text they send is presented in real-time in a
   readable way to the other participants even if they send
   simultaneously with other users and even when they make brief edit
   operations of their text by backspacing and correcting their text.

   Text is supposed to be human generated, by some text input means,
   such as typing on a keyboard or using speech-to-text technology.
   Occasional small cut-and-paste operations may appear even if that is
   not the initial purpose of real-time text.

   The real-time characteristics of real-time text is essential for the
   participants to be able to contribute to a conversation.  If the text
   is too much delayed from typing a letter to its presentation, then,
   in some conference situations, the opportunity to comment will be
   gone and someone else will grab the turn.  A delay of more than one
   second in such situations is an obstacle for good conversation.

2.  Overview of the two specified solutions and selection of method

   This section contains a brief introduction of the two methods
   specified in this document.

2.1.  The RTP-mixer based solution for multi-party aware endpoints

   This method specifies negotiated use of the RFC 4103 format for
   multi-party transmission in a single RTP stream.  The main purpose of
   this document is to specify a method for true multi-party real-time
   text mixing for multi-party aware endpoints that can be widely
   deployed.  The RTP-mixer based method makes use of the current format
   for real-time text in [RFC4103].  It is an update of RFC 4103 by a
   clarification on one way to use it in the multi-party situation.
   That is done by completing a negotiation for this kind of multi-party
   capability and by interleaving packets from different sources.  The
   source is indicated in the CSRC element in the RTP packets.  Specific
   considerations are made to be able to recover text after packet loss.

   The detailed procedures for the RTP-mixer based multi-party aware
   case are specified in Section 3.

   Please use [RFC4103] as reference when reading the specification.

2.2.  Mixing for multi-party unaware endpoints

   A method is also specified in this document for cases when the
   endpoint participating in a multi-party call does not itself
   implement any solution, or not the same, as the mixer.  The method
   requires the mixer to insert text dividers and readable labels and
   only send text from one source at a time until a suitable point
   appears for source change.  This solution is a fallback method with
   functional limitations.  It acts on the presentation level.

   A party acting as a mixer, which has not negotiated any method for
   true multi-party RTT handling, but negotiated a "text/red" or "text/
   t140" format in a session with a participant SHOULD in order to
   maintain interoperability, if nothing else is specified for the
   application, format transmitted text to that participant to be
   suitable to present on a multi-party unaware endpoint as further
   specified in Section 4.2.

2.3.  Offer/answer considerations

   RTP Payload for Text Conversation [RFC4103] specifies use of RTP
   [RFC3550], and a redundancy format "text/red" for increased
   robustness of real-time text transmission.  This document updates
   [RFC4103] by introducing a capability negotiation for handling multi-
   party real-time text, a way to indicate the source of transmitted
   text, and rules for efficient timing of the transmissions interleaved
   from different sources.

   The capability negotiation for the "RTP-mixer based multi-party
   method" is based on use of the SDP media attribute "rtt-mixer".

   Both parties SHALL indicate their capability in a session setup or
   modification, and evaluate the capability of the counterpart.

   The syntax is as follows:

   If any other method for RTP-based multi-party real-time text gets
   specified, it is assumed that it will be recognized by some specific
   SDP feature exchange.

   It is possible to both indicate capability for the RTP-mixer based
   method and another method.  An answer MUST NOT accept more than one

2.4.  Actions depending on capability negotiation result

   A transmitting party SHALL send text according to the RTP-mixer based
   multi-party method only when the negotiation for that method was
   successful and when it conveys text for another source.  In all other
   cases, the packets SHALL be populated and interpreted as for a two-
   party session.

   A party which has negotiated the "rtt-mixer" SDP media attribute MUST
   populate the CSRC-list and format the packets according to Section 3
   if it acts as an rtp-mixer and sends multi-party text.

   A party which has negotiated the "rtt-mixer" SDP media attribute MUST
   interpret the contents of the "CC" field, the CSRC-list and the
   packets according to Section 3 in received RTP packets in the
   corresponding RTP stream.

   A party which has not successfully completed the negotiation of the
   "rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved
   from different sources in the same RTP stream as specified in
   Section 3.  If the party is a mixer and did declare the "rtt-mixer"
   SDP media attribute, it SHOULD perform the procedure for multi-party
   unaware endpoints.  If the party is not a mixer, it SHOULD transmit
   according to [RFC4103].

3.  Details for the RTP-mixer based multi-party aware mixing method

3.1.  Use of fields in the RTP packets

   The CC field SHALL show the number of members in the CSRC list, which
   SHALL be one (1) in transmissions from a mixer when conveying text
   from other sources in a multi-party session, and otherwise 0.

   When text is conveyed by a mixer during a multi-party session, a CSRC
   list SHALL be included in the packet.  The single member in the CSRC-
   list SHALL contain the SSRC of the source of the T140blocks in the

   When redundancy is used, the RECOMMENDED level of redundancy is to
   use one primary and two redundant generations of T140blocks.  In some
   cases, a primary or redundant T140block is empty, but is still
   represented by a member in the redundancy header.

   From other aspects, the contents of the RTP packets are equal to what
   is specified in [RFC4103].

3.2.  Initial transmission of a BOM character

   As soon as a participant is known to participate in a session with
   another entity and is available for text reception, a Unicode BOM
   character SHALL be sent to it by the other entity according to the
   procedures in this section.  If the transmitter is a mixer, then the
   source of this character SHALL be indicated to be the mixer itself.

   Note that the BOM character SHALL be transmitted with the same
   redundancy procedures as any other text.

3.3.  Keep-alive

   After that, the transmitter SHALL send keep-alive traffic to the
   receiver(s) at regular intervals when no other traffic has occurred
   during that interval, if that is decided for the actual connection.
   It is RECOMMENDED to use the keep-alive solution from [RFC6263].  The
   consent check of [RFC7675] is a possible alternative if it is used
   anyway for other reasons.

3.4.  Transmission interval

   A "text/red" or "text/t140" transmitter in a mixer SHALL send packets
   distributed in time as long as there is something (new or redundant
   T140blocks) to transmit.  The maximum transmission interval SHALL
   then be 330 ms, when no other limitations cause a longer interval to
   be temporarily used.  It is RECOMMENDED to send the next packet to a
   receiver as soon as new text to that receiver is available, as long
   as the maximum character rate ("CPS") to the receiver is not exceeded
   during any 10 second interval.  The intention of these time intervals
   is to keep the latency low and network load limited while keeping a
   good protection against text loss in bursty packet loss conditions.
   The main purpose of the 330 ms interval is for timing of redundant
   transmission, when no new text from the same source is available.

   If the "CPS" value is reached, longer transmission intervals SHALL be
   applied and only part of the text queued for transmission sent at end
   of each transmission interval, until the transmission rate falls
   under the "CPS" value again.  See also Section 8

   For a transmitter not acting in a mixer, the transmission interval
   principles from [RFC4103] apply, and the transmission interval SHALL
   be 300 ms.

3.5.  Only one source per packet

   New text and redundant copies of earlier text from one source SHALL
   be transmitted in the same packet if available for transmission at
   the same time.  Text from different sources MUST NOT be transmitted
   in the same packet.

3.6.  Do not send received text to the originating source

   Text received by a mixer from a participant SHOULD NOT be included in
   transmission from the mixer to that participant, because the normal
   behavior of the endpoint is to present locally produced locally.

3.7.  Clean incoming text

   A mixer SHALL handle reception, recovery from packet loss, deletion
   of superfluous redundancy, marking of possible text loss and deletion
   of 'BOM' characters from each participant before queueing received
   text for transmission to receiving participants.

3.8.  Redundant transmission principles

   A transmitting party using redundancy SHALL send redundant
   repetitions of T140blocks already transmitted in earlier packets.

   The number of redundant generations of T140blocks to include in
   transmitted packets SHALL be deduced from the SDP negotiation.  It
   SHALL be set to the minimum of the number declared by the two parties
   negotiating a connection.  It is RECOMMENDED to declare and transmit
   one original and two redundant generations of the T140blocks, because
   that provides good protection against text loss in case of packet
   loss, and low overhead.

3.9.  Interleaving text from different sources

   When text from more than one source is available for transmission
   from a mixer, the mixer SHALL let the sources take turns in having
   their text transmitted.

   The source with the oldest text received in the mixer or oldest
   redundant text SHALL be next in turn to get all its available unsent
   text transmitted.  Any redundant repetitions of earlier transmitted
   text not yet sent the intended number of times SHALL be included as
   redundant retransmission in the transmission.

3.10.  Text placement in packets

   The mixer SHALL compose and transmit an RTP packet to a receiver when
   one of the following conditions has occurred:

   *  There is unsent text available for transmission to that receiver.

   *  330 ms has passed since already transmitted text was queued for
      transmission as redundant text.

   At time of transmission, the mixer SHALL populate the RTP packet with
   all T140blocks queued for transmission originating from the source in
   turn for transmission as long as this is not in conflict with the
   allowed number of characters per second ("CPS") or the maximum packet
   size.  In this way, the latency of the latest received text is kept
   low even in moments of simultaneous transmission from many sources.

   Redundant text SHALL also be included.  See Section 3.12

   The SSRC of the source SHALL be placed as the only member in the

   Note: The CSRC-list in an RTP packet only includes the participant
   whose text is included in text blocks.  It is not the same as the
   total list of participants in a conference.  With audio and video
   media, the CSRC-list would often contain all participants who are not
   muted whereas text participants that don't type are completely silent
   and thus are not represented in RTP packet CSRC-lists.

3.11.  Empty T140blocks

   If no unsent T140blocks were available for a source at the time of
   populating a packet, but T140blocks are available which have not yet
   been sent the full intended number of redundant transmissions, then
   the primary T140block for that source is composed of an empty
   T140block, and populated (without taking up any length) in a packet
   for transmission.  The corresponding SSRC SHALL be placed as usual in
   its place in the CSRC-list.

   The first packet in the session, the first after a source switch and
   the first after a pause SHALL be poulated with the available
   T140blocks for the source in turn to be sent as primary, and empty
   T140blocks for the agreed number of redundancy generations.

3.12.  Creation of the redundancy

   The primary T140block from a source in the latest transmitted packet
   is saved for populating the first redundant T140block for that source
   in next transmission of text from that source.  The first redundant
   T140block for that source from the latest transmission is saved for
   populating the second redundant T140block in next transmission of
   text from that source.

   Usually this is the level of redundancy used.  If a higher number of
   redundancy is negotiated, then the procedure SHALL be maintained
   until all available redundant levels of T140blocks are placed in the
   packet.  If a receiver has negotiated a lower number of "text/red"
   generations, then that level SHALL be the maximum used by the

   The T140blocks saved for transmission as redundant data are assigned
   a planned transmission time 330 ms after the current time, but SHOULD
   be transmitted earlier if new text for the same source gets in turn
   for transmission before that time.

3.13.  Timer offset fields

   The timestamp offset values SHALL be inserted in the redundancy
   header, with the time offset from the RTP timestamp in the packet
   when the corresponding T140block was sent as primary.

   The timestamp offsets are expressed in the same clock tick units as
   the RTP timestamp.

   The timestamp offset values for empty T140blocks have no relevance
   but SHOULD be assigned realistic values.

3.14.  Other RTP header fields

   The number of members in the CSRC list ( 0 or 1) SHALL be placed in
   the "CC" header field.  Only mixers place value 1 in the "CC" field.
   A value of "0" indicates that the source is the transmitting device
   itself and that the source is indicated by the SSRC field.  This
   value is used by endpoints, and by mixers sending data that it is
   source of itself.

   The current time SHALL be inserted in the timestamp.

   The SSRC of the mixer for the RTT session SHALL be inserted in the
   SSRC field of the RTP header.

   The M-bit SHALL be handled as specified in [RFC4103].

3.15.  Pause in transmission

   When there is no new T140block to transmit, and no redundant
   T140block that has not been retransmitted the intended number of
   times from any source, the transmission process SHALL be stopped
   until either new T140blocks arrive, or a keep-alive method calls for
   transmission of keep-alive packets.

3.16.  RTCP considerations

   A mixer SHALL send RTCP reports with SDES, CNAME and NAME information
   about the sources in the multi-party call.  This makes it possible
   for participants to compose a suitable label for text from each

   Integrity SHALL be considered when composing these fields.  They
   contain name and address information that may be sensitive to
   transmit in its entirety e.g. to unauthenticated participants.
   Similar considerations SHALL be taken as for other media.

3.17.  Reception of multi-party contents

   The "text/red" receiver included in an endpoint with presentation
   functions will receive RTP packets in the single stream from the
   mixer, and SHALL distribute the T140blocks for presentation in
   presentation areas for each source.  Other receiver roles, such as
   gateways or chained mixers are also feasible, and requires
   consideration if the stream shall just be forwarded, or distributed
   based on the different sources.

3.17.1.  Acting on the source of the packet contents

   If the "CC" field value of a received packet is 1, it indicates that
   the text is conveyed from a source indicated in the single member in
   the CSRC-list, and the receiver MUST act on the source according to
   its role.  If the CC value is 0, the source is indicated in the SSRC

3.17.2.  Detection and indication of possible text loss

   The RTP sequence numbers of the received packets SHALL be monitored
   for gaps and packets out of order.  If a sequence number gap appears
   and still exists after some defined short time for jitter resolution,
   the packets in the gap SHALL be regarded as lost.

   If it is known that only one source is active in the RTP session,
   then it is likely that a gap equal to or larger than the agreed
   number of redundancy generations (including the primary) causes text
   loss.  In that case a t140block SHALL be created with a marker for
   possible text loss [T140ad1] and assigned to the source and inserted
   in the reception buffer for that source.

   If it is known that more than one source is active in the RTP
   session, then it is not possible in general to evaluate if text was
   lost when packets were lost.  With two active sources and the
   recommended number of redundancy generations (3), it can take a gap
   of five consecutive lost packets until any text may be lost, but text
   loss can also appear if three non-consecutive packets are lost when
   they contained consecutive data from the same source.  A simple
   method to decide when there is risk for resulting text loss is to
   evaluate if three or more packets were lost within one second.  If
   this simple method is used, then a t140block SHOULD be created with a
   marker for possible text loss [T140ad1] and assigned to the SSRC of
   the transmitter as a general input from the mixer.

   Implementations MAY apply more refined methods for more reliable
   detection of if text was lost or not.  Any refined method SHALL
   prefer marking possible loss rather than not marking when it is
   uncertain if there was loss.

3.17.3.  Extracting text and handling recovery

   When applying the following procedures, the effects MUST be
   considered of possible timestamp wrap around and the RTP session
   possibly changing SSRC.

   When a packet is received in an RTP session using the packetization
   for multi-party aware endpoints, its T140blocks SHALL be extracted in
   the following way.  The description is adapted to the default
   redundancy case using the original and two redundant generations.

   The source SHALL be extracted from the CSRC-list if available,
   otherwise from the SSRC.

   If the received packet is the first packet received from the source,
   then all T140blocks in the packet SHALL be retrieved and assigned to
   a receive buffer for the source beginning with the second generation
   redundancy, continuing with the first generation redundancy and
   finally the primary.

   Note: The normal case is that in the first packet, only the primary
   data has contents.  The redundant data has contents in the first
   received packet from a source only after initial packet loss.

   If the packet is not the first packet from a source, then if the
   second generation redundant data is available, its timestamp SHALL be
   created by subtracting its timestamp offset from the RTP timestamp.
   If the resulting timestamp is later than the latest retrieved data
   from the same source, then the redundant data SHALL be retrieved and
   appended to the receive buffer.  The process SHALL be continued in
   the same way for the first generation redundant data.  After that,
   the primary data SHALL be retrieved from the packet and appended to
   the receive buffer for the source.

3.17.4.  Delete 'BOM'

   Unicode character 'BOM' is used as a start indication and sometimes
   used as a filler or keep alive by transmission implementations.
   These SHALL be deleted after extraction from received packets.

3.18.  Performance considerations

   This solution has good performance with low text delays as long as
   the sum of characters per second during any 10 second interval sent
   from a number of simultaneously sending participants to a receiving
   participant does not reach the 'CPS' value.  At higher numbers of
   characters per second sent, a jerkiness is visible in the
   presentation of text.  The solution is therefore suitable for
   emergency service use, relay service use, and small or well-managed
   larger multimedia conferences.  Only in large unmanaged conferences
   with a high number of participants there may on very rare occasions
   appear situations when many participants happen to send text
   simultaneously, resulting in unpleasantly jerky presentation of text
   from each sending participant.  It should be noted that it is only
   the number of users sending text within the same moment that causes
   jerkiness, not the total number of users with RTT capability.

3.19.  Security for session control and media

   Security SHOULD be applied when possible regarding the capabilities
   of the participating devices by use of SIP over TLS by default
   according to [RFC5630] section 3.1.3 on session control level and by
   default using DTLS-SRTP [RFC5764] on media level.  In applications
   where legacy endpoints without security may exist, a negotiation
   SHOULD be performed to decide if security by encryption on media
   level will be applied.  If no other security solution is mandated for
   the application, then OSRTP [RFC8643] is a suitable method be applied
   to negotiate SRTP media security with DTLS.  Most SDP examples below
   are for simplicity expressed without the security additions.  The
   principles (but not all details) for applying DTLS-SRTP [RFC5764]
   security is shown in a couple of the following examples.

   This document contains two mixing procedures which imply different
   security levels.  The mixing for conference-unaware endpoints has
   lower security level than the mixing method for conference-aware
   endpoints, because there may be an opportunity for a malicious mixer
   or a middleman to masquerade the source labels accompanying the text
   streams in text format.  This is especially true if support of un-
   encrypted SIP and media is supported because of lack of such support
   in the target endpoints.  However, the mixing for conference-aware
   endpoints as specified here also requires that the mixer can be
   trusted.  End to end encryption would require further work and could
   be based on WebRTC as specified in Section 1.2.

3.20.  SDP offer/answer examples

   This section shows some examples of SDP for session negotiation of
   the real-time text media in SIP sessions.  Audio is usually provided
   in the same session, and sometimes also video.  The examples only
   show the part of importance for the real-time text media.  The
   examples relate to the single RTP stream mixing for multi-party aware
   endpoints and for multi-party unaware endpoints.

   Note: Multi-party RTT MAY also be provided through other methods,
   e.g. by a Selective Forwarding Middlebox (SFM).  In that case, the
   SDP of the offer will include something specific for that method, and
   an answer acknowledging the use of that method would accept it by
   something specific included in the SDP.  The offer may contain also
   the "rtt-mixer" SDP media attribute for the main RTT media when the
   offeror has capability for both multi-party methods, while an answer,
   selecting to use SFM will not include the "rtt-mixer" SDP media

     Offer example for "text/red" format and multi-party support:

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

      Answer example from a multi-party capable device
           m=text 14000 RTP/AVP 100 98
           a=rtpmap:98 t140/1000
           a=rtpmap:100 red/1000
           a=fmtp:100 98/98/98

      Offer example for "text/red" format including multi-party
      and security:
            a=fingerprint: (fingerprint1)
            m=text 11000 RTP/AVP 100 98
            a=rtpmap:98 t140/1000
            a=rtpmap:100 red/1000
            a=fmtp:100 98/98/98

   The "fingerprint" is sufficient to offer DTLS-SRTP, with the media
   line still indicating RTP/AVP.

   Note: For brevity, the entire value of the SDP fingerprint attribute
   is not shown in this and the following example.

       Answer example from a multi-party capable device with security
            a=fingerprint: (fingerprint2)
            m=text 16000 RTP/AVP 100 98
            a=rtpmap:98 t140/1000
            a=rtpmap:100 red/1000
            a=fmtp:100 98/98/98

   With the "fingerprint" the device acknowledges use of SRTP/DTLS.

     Answer example from a multi-party unaware device that also
     does not support security:

           m=text 12000 RTP/AVP 100 98
           a=rtpmap:98 t140/1000
           a=rtpmap:100 red/1000
           a=fmtp:100 98/98/98

3.21.  Packet sequence example from interleaved transmission

   This example shows a symbolic flow of packets from a mixer including
   loss and recovery.  The sequence includes interleaved transmission of
   text from two RTT sources A and B.  P indicates primary data.  R1 is
   first redundant generation data and R2 is the second redundant
   generation data.  A1, B1, A2 etc are text chunks (T140blocks)
   received from the respective sources and sent on to the receiver by
   the mixer.  X indicates dropped packet between the mixer and a
   receiver.  The session is assumed to use original and two redundant
   generations of RTT.

     |Seq no 101, Time=20400 |
     |CC=1                   |
     |CSRC list A            |
     |R2: A1, Offset=600     |
     |R1: A2, Offset=300     |
     |P:  A3                 |

   Assuming that earlier packets ( with text A1 and A2) were received in
   sequence, text A3 is received from packet 101 and assigned to
   reception area A.  The mixer is now assumed to have received text
   from source B 100 ms after packet 101 and will send that text.
   Transmission of A2 and A3 as redundancy is planned for 330 ms after
   packet 101 if no new text from A is ready to be sent before that.

     |Seq no 102, Time=20500 |
     |CC=1                   |
     |CSRC list B            |
     |R2  Empty, Offset=600  |
     |R1: Empty, Offset=300  |
     |P:  B1                 |
     Packet 102 is received.
     B1 is retrieved from this packet. Redundant transmission of
     B1 is planned 330 ms after packet 102.

     X Seq no 103, Timer=20730|
     X CC=1                   |
     X CSRC list A            |
     X R2: A2, Offset=630     |
     X R1: A3, Offset=330     |
     X P:  Empty              |
     Packet 103 is assumed to be lost due to network problems.
     It contains redundancy for A. Sending A3 as second level
     redundancy is planned for 330 ms after packet 103.

     X Seq no 104, Timer=20830|
     X CC=1                   |
     X CSRC list B            |
     X R2: Empty, Offset=600  |
     X R1: B1, Offset=300     |
     X P:  B2                 |
     Packet 104 contains text from B, including new B2 and
     redundant B1. It is assumed dropped in network
     The mixer has A3 redundancy to send but no new text
     appears from A and therefore the redundancy is sent
     330 ms after the previous packet with text from A.

     | Seq no 105, Timer=21060|
     | CC=1                   |
     | CSRC list A            |
     | R2: A3, Offset=660     |
     | R1: Empty, Offset=330  |
     | P:  Empty              |
     Packet 105 is received.
     A gap for lost 103, and 104 is detected.
     Assume that no other loss was detected the last second.
     Then it can be concluded that nothing was totally lost.

     R2 is checked. Its original time was 21040-660=20400.
     A packet with text from A was received with that
     timestamp, so nothing needs to be recovered.

     B1 and B2 still needs to be transmitted as redundancy.
     This is planned 330 ms after packet 105. That
     would be at 21150.

     |Seq no 106, Timer=21160|
     |CC=1                   |
     |CSRC list B            |
     | R2: B1, Offset=660    |
     | R1: B2, Offset=330    |
     | P:  Empty             |

   Packet 106 is received.

   The second level redundancy in packet 106 is B1 and has timestamp
   offset 660 ms.  The timestamp of packet 106 minus 660 is 20500 which
   is the timestamp of packet 102 THAT was received.  So B1 does not
   need to be retrieved.  The first level redundancy in packet 106 has
   offset 330.  The timestamp of packet 106 minus 330 is 20830.  That is
   later than the latest received packet with source B.  Therefore B2 is
   retrieved and assigned to the input buffer for source B.  No primary
   is available in packet 106.

   After this sequence, A3 and B1 and B2 have been received.  In this
   case no text was lost.

3.22.  Maximum character rate "CPS"

   The default maximum rate of reception of "text/t140" real-time text
   is in [RFC4103] specified to be 30 characters per second.  The value
   MAY be modified in the "CPS" parameter of the FMTP attribute in the
   media section for the "text/t140" media.  A mixer combining real-time
   text from a number of sources may occasionally have a higher combined
   flow of text coming from the sources.  Endpoints SHOULD therefore
   specify a suitable higher value for the "CPS" parameter,
   corresponding to its real reception capability.  A value for "CPS" of
   90 SHALL be the default for the "text/t140" stream in the "text/red"
   format when multi-party real-time text is negotiated.  See [RFC4103]
   for the format and use of the "CPS" parameter.  The same rules apply
   for the multi-party case except for the default value.

4.  Presentation level considerations

   "Protocol for multimedia application text conversation" [T140]
   provides the presentation level requirements for the [RFC4103]
   transport.  Functions for erasure and other formatting functions and
   are specified in [T140] which has the following general statement for
   the presentation:

   "The display of text from the members of the conversation should 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."

   Strict application of [T140] is of essence for the interoperability
   of real-time text implementations and to fulfill the intention that
   the session participants have the same information of the text
   contents of the conversation without necessarily having the exact
   same layout of the conversation.

   [T140] specifies a set of presentation control codes to include in
   the stream.  Some of them are optional.  Implementations MUST be able
   to ignore optional control codes that they do not support.

   There is no strict "message" concept in real-time text.  The Unicode
   Line Separator character SHALL be used as a separator allowing a part
   of received text to be grouped in presentation.  The characters
   "CRLF" may be used by other implementations as replacement for Line
   Separator.  The "CRLF" combination SHALL be erased by just one
   erasing action, just as the Line Separator.  Presentation functions
   are allowed to group text for presentation in smaller groups than the
   line separators imply and present such groups with source indication
   together with text groups from other sources (see the following
   presentation examples).  Erasure has no specific limit by any
   delimiter in the text stream.

4.1.  Presentation by multi-party aware endpoints

   A multi-party aware receiving party, presenting real-time text MUST
   separate text from different sources and present them in separate
   presentation fields.  The receiving party MAY separate presentation
   of parts of text from a source in readable groups based on other
   criteria than line separator and merge these groups in the
   presentation area when it benefits the user to most easily find and
   read text from the different participants.  The criteria MAY e.g. be
   a received comma, full stop, or other phrase delimiters, or a long

   When text is received from multiple original sources, the
   presentation SHALL provide a view where text is added in multiple
   presentation fields.

   If the presentation presents text from different sources in one
   common area, the presenting endpoint SHOULD insert text from the
   local user ended at suitable points merged with received text to
   indicate the relative timing for when the text groups were completed.
   In this presentation mode, the receiving endpoint SHALL present the
   source of the different groups of text.  This presentation style is
   called the "chat" style here and provides a possibility to follow
   text arriving from multiple parties and the approximate relative time
   that text is received related to text from the local user.

   A view of a three-party RTT call in chat style is shown in this
   example .

                |                                              |^|
                |[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                           |-|
                | of course, I underst                           |

   Figure 3: Example of a three-party RTT call presented in chat style
   seen at participant 'Alice's endpoint.

   Other presentation styles than the chat style MAY be arranged.

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

   |       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 4: An example of a coordinated column-view of a three-party
   session with entries ordered vertically in approximate time-order.

4.2.  Multi-party mixing for multi-party unaware endpoints

   When the mixer has indicated RTT multi-party capability in an SDP
   negotiation, but the multi-party capability negotiation fails with an
   endpoint, then the agreed "text/red" or "text/t140" format SHALL be
   used and the mixer SHOULD compose a best-effort presentation of
   multi-party real-time text in one stream intended to be presented by
   an endpoint with no multi-party awareness, when that is desired in
   the actual implementation.  The following specifies a procedure which
   MAY be applied in that situation.

   This presentation format has functional limitations and SHOULD be
   used only to enable participation in multi-party calls by legacy
   deployed endpoints implementing only RFC 4103 without any multi-party
   extensions specified in this document.

   The principles and procedures below do not specify any new protocol
   elements.  They are instead composed from the information in [T140]
   and an ambition to provide a best effort presentation on an endpoint
   which has functions only for two-party calls.

   The mixer mixing for multi-party unaware endpoints SHALL compose a
   simulated limited multi-party RTT view suitable for presentation in
   one presentation area.  The mixer SHALL group text in suitable groups
   and prepare for presentation of them by inserting a new line between
   them if the transmitted text did not already end with a new line.  A
   presentable label SHALL be composed and sent for the source initially
   in the session and after each source switch.  With this procedure the
   time for switching from transmission of text from one source to
   transmission of text from another source is depending on the actions
   of the users.  In order to expedite source switch, a user can for
   example end its turn with a new line.

4.2.1.  Actions by the mixer at reception from the call participants

   When text is received by the mixer from the different participants,
   the mixer SHALL recover text from redundancy if any packets are lost.
   The mark for lost text [T140ad1] SHALL be inserted in the stream if
   unrecoverable loss appears.  Any Unicode "BOM" characters, possibly
   used for keep-alive SHALL be deleted.  The time of creation of text
   (retrieved from the RTP timestamp) SHALL be stored together with the
   received text from each source in queues for transmission to the
   recipients in order to be able to evaluate text loss.

4.2.2.  Actions by the mixer for transmission to the recipients

   The following procedure SHALL be applied for each multi-party unaware
   recipient of multi-party text from the mixer.

   The text for transmission SHALL be formatted by the mixer for each
   receiving user for presentation in one single presentation area.
   Text received from a participant SHOULD NOT be included in
   transmission to that participant because it is usually presented
   locally at transmission time.  When there is text available for
   transmission from the mixer to a receiving party from more than one
   participant, the mixer SHALL switch between transmission of text from
   the different sources at suitable points in the transmitted stream.

   When switching source, the mixer SHALL insert a line separator if the
   already transmitted text did not end with a new line (line separator
   or CRLF).  A label SHALL be composed from information in the CNAME
   and NAME fields in RTCP reports from the participant to have its text
   transmitted, or from other session information for that user.  The
   label SHALL be delimited by suitable characters (e.g. '[ ]') and
   transmitted.  The CSRC SHALL indicate the selected source.  Then text
   from that selected participant SHALL be transmitted until a new
   suitable point for switching source is reached.

   Integrity considerations SHALL be taken when composing the label.

   Seeking a suitable point for switching source SHALL be done when
   there is older text waiting for transmission from any party than the
   age of the last transmitted text.  Suitable points for switching are:

   *  A completed phrase ended by comma

   *  A completed sentence

   *  A new line (line separator or CRLF)

   *  A long pause (e.g. > 10 seconds) in received text from the
      currently transmitted source

   *  If text from one participant has been transmitted with text from
      other sources waiting for transmission for a long time (e.g. > 1
      minute) and none of the other suitable points for switching has
      occurred, a source switch MAY be forced by the mixer at next word
      delimiter, and also if even a word delimiter does not occur within
      a time (e.g. 15 seconds) after the scan for word delimiter

   When switching source, the source which has the oldest text in queue
   SHALL be selected to be transmitted.  A character display count SHALL
   be maintained for the currently transmitted source, starting at zero
   after the label is transmitted for the currently transmitted source.

   The status SHALL be maintained for the latest control code for Select
   Graphic Rendition (SGR) from each source.  If there is an SGR code
   stored as the status for the current source before the source switch
   is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B
   0000 006D] after the new line and before the new label during a
   source switch.  See SGR below for an explanation.  This transmission
   does not influence the display count.

   If there is an SGR code stored for the new source after the source
   switch, that SGR code SHALL be transmitted to the recipient before
   the label.  This transmission does not influence the display count.

4.2.3.  Actions on transmission of text

   Text from a source sent to the recipient SHALL increase the display
   count by one per transmitted character.

4.2.4.  Actions on transmission of control codes

   The following control codes specified by T.140 require specific
   actions.  They SHALL cause specific considerations in the mixer.
   Note that the codes presented here are expressed in UCS-16, while
   transmission is made in UTF-8 transform of these codes.

   BEL 0007 Bell  Alert in session, provides for alerting during an
      active session.  The display count SHALL NOT be altered.

   NEW LINE 2028  Line separator.  Check and perform a source switch if
      appropriate.  Increase display count by 1.

   CR LF 000D 000A  A supported, but not preferred way of requesting a
      new line.  Check and perform a source switch if appropriate.
      Increase display count by 1.

   INT ESC 0061  Interrupt (used to initiate mode negotiation
      procedure).  The display count SHALL NOT be altered.

   SGR 009B Ps 006D  Select graphic rendition.  Ps is rendition
      parameters specified in ISO 6429.  The display count SHALL NOT be
      altered.  The SGR code SHOULD be stored for the current source.

   SOS 0098  Start of string, used as a general protocol element
      introducer, followed by a maximum 256 bytes string and the ST.
      The display count SHALL NOT be altered.

   ST 009C  String terminator, end of SOS string.  The display count
      SHALL NOT be altered.

   ESC 001B  Escape - used in control strings.  The display count SHALL
      NOT be altered for the complete escape code.

   Byte order mark "BOM" (U+FEFF)  "Zero width, no break space", used
      for synchronization and keep-alive, SHALL be deleted from incoming
      streams.  It SHALL also be sent first after session establishment
      to the recipient.  The display count SHALL NOT be altered.

   Missing text mark (U+FFFD)  "Replacement character", represented as a
      question mark in a rhombus, or if that is not feasible, replaced
      by an apostrophe ', marks place in stream of possible text loss.
      This mark SHALL be inserted by the reception procedure in case of
      unrecoverable loss of packets.  The display count SHALL be
      increased by one when sent as for any other character.

   SGR  If a control code for selecting graphic rendition (SGR), other
      than reset of the graphic rendition (SGR 0) is sent to a
      recipient, that control code SHALL also be stored as status for
      the source in the storage for SGR status.  If a reset graphic
      rendition (SGR 0) originated from a source is sent, then the SGR
      status storage for that source SHALL be cleared.  The display
      count SHALL NOT be increased.

   BS (U+0008)  Back Space, intended to erase the last entered character
      by a source.  Erasure by backspace cannot always be performed as
      the erasing party intended.  If an erasing action erases all text
      up to the end of the leading label after a source switch, then the
      mixer MUST NOT transmit more backspaces.  Instead it is
      RECOMMENDED that a letter "X" is inserted in the text stream for
      each backspace as an indication of the intent to erase more.  A
      new line is usually coded by a Line Separator, but the character
      combination "CRLF" MAY be used instead.  Erasure of a new line is
      in both cases done by just one erasing action (Backspace).  If the
      display count has a positive value it SHALL be decreased by one
      when the BS is sent.  If the display count is at zero, it SHALL
      NOT not altered.

4.2.5.  Packet transmission

   A mixer transmitting to a multi-party unaware terminal SHALL send
   primary data only from one source per packet.  The SSRC SHALL be the
   SSRC of the mixer.  The CSRC list SHALL contain one member and be the
   SSRC of the source of the primary data.

4.2.6.  Functional limitations

   When a multi-party unaware endpoint presents a conversation in one
   display area in a chat style, it inserts source indications for
   remote text and local user text as they are merged in completed text
   groups.  When an endpoint using this layout receives and presents
   text mixed for multi-party unaware endpoints, there will be two
   levels of source indicators for the received text; one generated by
   the mixer and inserted in a label after each source switch, and
   another generated by the receiving endpoint and inserted after each
   switch between local and remote source in the presentation area.
   This will waste display space and look inconsistent to the reader.

   New text can be presented only from one source at a time.  Switch of
   source to be presented takes place at suitable places in the text,
   such as end of phrase, end of sentence, line separator and
   inactivity.  Therefore the time to switch to present waiting text
   from other sources may become long and will vary and depend on the
   actions of the currently presented source.

   Erasure can only be done up to the latest source switch.  If a user
   tries to erase more text, the erasing actions will be presented as
   letter X after the label.

   Text loss because of network errors may hit the label between entries
   from different parties, causing risk for misunderstanding from which
   source a piece of text is.

   These facts make it strongly RECOMMENDED to implement multi-party
   awareness in RTT endpoints.  The use of the mixing method for multi-
   party-unaware endpoints should be left for use with endpoints which
   are impossible to upgrade to become multi-party aware.

4.2.7.  Example views of presentation on multi-party unaware endpoints

   The following pictures are examples of the view on a participant's
   display for the multi-party-unaware case.

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

   Figure 5: Alice who has a conference-unaware client is receiving the
   multi-party real-time text in a single-stream.

   This figure shows how a coordinated column view MAY be presented on
   Alice's device. device in a view with two-columns.  The mixer inserts labels
   to show how the sources alternate in the column with received text.
   The mixer alternates between the sources at suitable points in the
   text exchange so that text entries from each party can be
   conveniently read.

    |                                              |^|
    |(Alice) Hi, Alice here.                       |-|
    |                                              | |
    |(mix)[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.| |
    |                                              | |
    |(mix)[Bob] And I on Wednesday evening.        | |
    |                                              | |
    |[Eve] we can have dinner and then walk        | |
    |                                              | |
    |[Eve] But I need to be back to                | |
    |    the hotel by 11 because I need            | |
    |                                              |-|
    | of course, I underst                           |

   Figure 6: An example of a view of the multi-party unaware
   presentation in chat style.  Alice is the local user.

   In this view, there is a tradition in receiving applications to
   include a label showing the source of the text, here shown with
   parenthesis "()".  The mixer also inserts source labels for the
   multi-party call participants, here shown with brackets "[]".

5.  Relation to Conference Control

5.1.  Use with SIP centralized conferencing framework

   The SIP conferencing framework, mainly specified in [RFC4353],
   [RFC4579] and [RFC4575] is suitable for coordinating sessions
   including multi-party RTT.  The RTT stream between the mixer and a
   participant is one and the same during the conference.  Participants
   get announced by notifications when participants are joining or
   leaving, and further user information may be provided.  The SSRC of
   the text to expect from joined users MAY be included in a
   notification.  The notifications MAY be used both for security
   purposes and for translation to a label for presentation to other

5.2.  Conference control

   In managed conferences, control of the real-time text media SHOULD be
   provided in the same way as other for media, e.g. for muting and
   unmuting by the direction attributes in SDP [RFC8866].

   Note that floor control functions may be of value for RTT users as
   well as for users of other media in a conference.

6.  Gateway Considerations

6.1.  Gateway considerations with Textphones (e.g.  TTYs).

   Multi-party RTT sessions may involve gateways of different kinds.
   Gateways involved in setting up sessions SHALL correctly reflect the
   multi-party capability or unawareness of the combination of the
   gateway and the remote endpoint beyond the gateway.

   One case that may occur is a gateway to PSTN for communication with
   textphones (e.g.  TTYs).  Textphones are limited devices with no
   multi-party awareness, and it SHOULD therefore be suitable for the
   gateway to not indicate multi-party awareness for that case.  Another
   solution is that the gateway indicates multi-party capability towards
   the mixer, and includes the multi-party mixer function for multi-
   party unaware endpoints itself.  This solution makes it possible to
   make adaptations for the functional limitations of the textphone

   More information on gateways to textphones (TTYs) is found in

6.2.  Gateway considerations with WebRTC.

   Gateway operation to real-time text in WebRTC may also be required.
   In WebRTC, RTT is specified in [RFC8865].

   A multi-party bridge may have functionality for communicating by RTT
   both in RTP streams with RTT and WebRTC T.140 data channels.  Other
   configurations may consist of a multi-party bridge with either
   technology for RTT transport and a separate gateway for conversion of
   the text communication streams between RTP and T.140 data channel.

   In WebRTC, it is assumed that for a multi-party session, one T.140
   data channel is established for each source from a gateway or bridge
   to each participant.  Each participant also has a data channel with
   two-way connection with the gateway or bridge.

   The t140 channel used both ways is for text from the WebRTC user and
   from the bridge or gateway itself to the WebRTC user.  The label
   parameter of this t140 channel is used as NAME field in RTCP to
   participants on the RTP side.  The other t140 channels are only for
   text from other participants to the WebRTC user.

   When a new participant has entered the session with RTP transport of
   RTT, a new T.140 channel SHOULD be established to WebRTC users with
   the label parameter composed from the NAME field in RTCP on the RTP

   When a new participant has entered the multi-party session with RTT
   transport in a WebRTC T.140 data channel, the new participant SHOULD
   be announced by a notification to RTP users.  The label parameter
   from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP
   side, or other available session information.

   When a participant on the RTP side disappears, the corresponding
   T.140 data channel(s) SHOULD be closed.

   When a WebRTC user of T.140 data channels disconnects from the mixer,
   the corresponding RTP streams or sources in an RTP-mixed stream
   SHOULD be closed.

   T.140 data channels MAY be opened and closed by negotiation or
   renegotiation of the session or by any other valid means as specified
   in section 1 of [RFC8865].

7.  Updates to RFC 4103

   This document updates [RFC4103] by introducing an SDP media attribute
   "rtt-mixer" for negotiation of multi-party mixing capability with the
   [RFC4103] format, and by specifying the rules for packets when multi-
   party capability is negotiated and in use.

8.  Congestion considerations

   The congestion considerations and recommended actions from [RFC4103]
   are valid also in multi-party situations.

   The first action in case of congestion SHALL be to temporarily
   increase the transmission interval up to two seconds.

   If the very unlikely situation appears that many participants in a
   conference send text simultaneously, simultaneously for a long period, a delay will may
   build up for presentation of text at the receivers because of if the limitation
   in characters per second("CPS") to be transmitted to the participants. participants
   is exceeded.  More delay than 7 seconds can cause confusion in the
   session.  It is therefore RECOMMENDED that an RTP-mixer based mixer
   discards such text in excess and inserts a general indication of
   possible text loss [T140ad1] in the session.  If the main text
   contributor is indicated in any way, the mixer MAY avoid deleting
   text from that participant.  It should however be noted that human
   creation of text normally contains pauses, when the transmission can
   catch up, so that the transmission overload situations are expected
   to be very rare.

9.  Acknowledgements

   James Hamlin for format and performance aspects.

10.  IANA Considerations

10.1.  Registration of the "rtt-mixer" SDP media attribute

   [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
   RFC number of this document.]
   IANA is asked to register the new SDP attribute "rtt-mixer".

   Contact name:  IESG

   Contact email:  iesg@ietf.org

   Attribute name:  rtt-mixer

   Attribute semantics:  See RFCXXXX Section 2.3

   Attribute value:  none

   Usage level:  media

   Purpose:  Indicate support by mixer and endpoint of multi-party
      mixing for real-time text transmission, using a common RTP-stream
      for transmission of text from a number of sources mixed with one
      source at a time and the source indicated in a single CSRC-list

   Charset Dependent:  no

   O/A procedure:  See RFCXXXX Section 2.3

   Mux Category:  normal

   Reference:  RFCXXXX

11.  Security Considerations

   The RTP-mixer model requires the mixer to be allowed to decrypt, pack
   and encrypt secured text from the conference participants.  Therefore
   the mixer needs to be trusted.  This is similar to the situation for
   central mixers of audio and video.

   The requirement to transfer information about the user in RTCP
   reports in SDES, CNAME and NAME fields, and in conference
   notifications, for creation of labels may have privacy concerns as
   already stated in RFC 3550 [RFC3550], and may be restricted for
   privacy reasons.  The receiving user will then get a more symbolic
   label for the source.

   Participants with malicious intentions may appear and e.g. disturb
   the multi-party session by a continuous flow of text, or masquerade
   as text from other participants.  Counteractions should be to require
   secure signaling, media and authentication, and to provide higher
   level conference functions e.g. for blocking and expelling

   Further security considerations specific for this application are
   specified in section Section 3.19.

12.  Change history

   [RFC Editor: Please remove this section prior to publication.]

12.1.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15

   Actions on review comments from Jurgen Schonwalder:

   A bit more about congestion situations and that they are expected to
   be very rare.

   Explanation of differences in security between the conference-aware
   and the conference-unaware case added in security section.

   Presentation examples with suource labels made less confusing, and

   Reference to T.140 inserted at first mentioning of T.140.

   Reference to RFC 8825 inserted to explain WebRTC

   Nit in wording in terminology section adjusted.

12.2.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14

   Changes from comments by Murray Cucherawy during AD review.

   Many SHOULD in section 4.2 on multi-party unaware mixing changed to
   SHALL, and the whole section instead specified to be optional
   depending on the application.

   Some SHOULD in section 3 either explained or changed to SHALL.

   In order to have explainable conditions behind SHOULDs, the
   transmission interval in 3.4 is changed to as soon as text is
   available as a main principle.  The call participants send with 300
   ms interval so that will create realistic load conditions anyway.


12.3.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13

   Changed year to 2021.

   Changed reference to draft on RTT in WebRTC to recently published RFC

   Changed label brackets in example from "[]" to "()" to avoid nits

   Changed reference "RFC 4566" to recently published "RFC 8866"


12.4.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12

   Changes according to responses on comments from Brian Rosen in
   Avtcore list on 2020-12-05 and -06.

   Changes according to responses to comments by Bernard Aboba in
   avtcore list 2020-12-06.

   Introduction of an optiona RTP multi-stream mixing method for further
   study as proposed by Bernard Aboba.

   Changes clarifying how to open and close T.140 data channels included
   in 6.2 after comments by Lorenzo Miniero.

   Changes to satisfy nits check.  Some "not" changed to "NOT" in
   normative wording combinations.  Some lower case normative words
   changed to upper case.  A normative reference deleted from the
   abstract.  Two informative documents moved from normative references
   to informative references.


12.5.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11

   Timestamps and timestamp offsets added to the packet examples in
   section 3.23, and the description corrected.

   A number of minor corrections added in sections 3.10 - 3.23.


12.6.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10

   The packet composition was modified for interleaving packets from
   different sources.

   The packet reception was modified for the new interleaving method.

   The packet sequence examples was adjusted for the new interleaving

   Modifications according to responses to Brian Rosen of 2020-11-03


12.7.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09

   Changed name on the SDP media attribute to "rtt-mixer"
   Restructure of section 2 for balance between aware and unaware cases.

   Moved conference control to own section.

   Improved clarification of recovery and loss in the packet sequence

   A number of editorial corrections and improvements.


12.8.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08

   Deleted the method requiring a new packet format "text/rex" because
   of the longer standardization and implementation period it needs.

   Focus on use of RFC 4103 text/red format with shorter transmission
   interval, and source indicated in CSRC.


12.9.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07

   Added a method based on the "text/red" format and single source per
   packet, negotiated by the "rtt-mixer" SDP attribute.

   Added reasoning and recommendation about indication of loss.

   The highest number of sources in one packet is 15, not 16.  Changed.

   Added in information on update to RFC 4103 that RFC 4103 explicitly
   allows addition of FEC method.  The redundancy is a kind of forward
   error correction..


12.10.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06

   Improved definitions list format.

   The format of the media subtype parameters is made to match the

   The mapping of media subtype parameters to SDP is included.

   The "CPS" parameter belongs to the t140 subtype and does not need to
   be registered here.


12.11.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05

   nomenclature and editorial improvements

   "this document" used consistently to refer to this document.


12.12.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04

   'Redundancy header' renamed to 'data header'.

   More clarifications added.

   Language and figure number corrections.


12.13.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03

   Mention possible need to mute and raise hands as for other media.
   ---done ----

   Make sure that use in two-party calls is also possible and explained.
   - may need more wording -

   Clarify the RTT is often used together with other media. --done--

   Tell that text mixing is N-1.  A users own text is not received in
   the mix. -done-

   In 3. correct the interval to: A "text/rex" transmitter SHOULD send
   packets distributed in time as long as there is something (new or
   redundant T140blocks) to transmit.  The maximum transmission interval
   SHOULD then be 300 ms.  It is RECOMMENDED to send a packet to a
   receiver as soon as new text to that receiver is available, as long
   as the time after the latest sent packet to the same receiver is more
   than 150 ms, and also the maximum character rate to the receiver is
   not exceeded.  The intention is to keep the latency low while keeping
   a good protection against text loss in bursty packet loss conditions.

   In 1.3 say that the format is used both ways. -done-

   In 13.1 change presentation area to presentation field so that reader
   does not think it shall be totally separated. -done-

   In Performance and intro, tell the performance in number of
   simultaneous sending users and introduced delay 16, 150 vs
   requirements 5 vs 500. -done --

   Clarify redundancy level per connection.  -done-

   Timestamp also for the last data header.  To make it possible for all
   text to have time offset as for transmission from the source.  Make
   that header equal to the others. -done-

   Mixer always use the CSRC list, even for its own BOM. -done-
   Combine all talk about transmission interval (300 ms vs when text has
   arrived) in section 3 in one paragraph or close to each other. -done-

   Documents the goal of good performance with low delay for 5
   simultaneous typers in the introduction. -done-

   Describe better that only primary text shall be sent on to receivers.
   Redundancy and loss must be resolved by the mixer. -done-


12.14.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02

   SDP and better description and visibility of security by OSRTP RFC
   8634 needed.

   The description of gatewaying to WebRTC extended.

   The description of the data header in the packet is improved.


12.15.  Changes to draft-ietf-avtcore-multi-party-rtt-mix-01

   2,5,6 More efficient format "text/rex" introduced and attribute
   a=rtt-mix deleted.

   3.  Brief about use of OSRTP for security included- More needed.

   4.  Brief motivation for the solution and why not rtp-translator is
   used added to intro.

   7.  More limitations for the multi-party unaware mixing method

   8.  Updates to RFC 4102 and 4103 more clearly expressed.

   9.  Gateway to WebRTC started.  More needed.


12.16.  Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03
        to draft-ietf-avtcore-multi-party-rtt-mix-00

   Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00

   Replaced CDATA in IANA registration table with better coding.

   Converted to xml2rfc version 3.


12.17.  Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02
        to -03

   Changed company and e-mail of the author.

   Changed title to "RTP-mixer formatting of multi-party Real-time text"
   to better match contents.

   Check and modification where needed of use of RFC 2119 words SHALL

   More about the CC value in sections on transmitters and receivers so
   that 1-to-1 sessions do not use the mixer format.

   Enhanced section on presentation for multi-party-unaware endpoints

   A paragraph recommending CPS=150 inserted in the performance section.


12.18.  Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01
        to -02

   In Abstract and 1.  Introduction: Introduced wording about regulatory

   In section 5: The transmission interval is decreased to 100 ms when
   there is text from more than one source to transmit.

   In section 11 about SDP negotiation, a SHOULD-requirement is
   introduced that the mixer should make a mix for multi-party unaware
   endpoints if the negotiation is not successful.  And a reference to a
   later chapter about it.

   The presentation considerations chapter 14 is extended with more
   information about presentation on multi-party aware endpoints, and a
   new section on the multi-party unaware mixing with low functionality
   but SHOULD a be implemented in mixers.  Presentation examples are

   A short chapter 15 on gateway considerations is introduced.

   Clarification about the text/t140 format included in chapter 10.

   This sentence added to the chapter 10 about use without redundancy.
   "The text/red format SHOULD be used unless some other protection
   against packet loss is utilized, for example a reliable network or

   Note about deviation from RFC 2198 added in chapter 4.

   In chapter 9.  "Use with SIP centralized conferencing framework" the
   following note is inserted: Note: The CSRC-list in an RTP packet only
   includes participants who's text is included in one or more text
   blocks.  It is not the same as the list of participants in a
   conference.  With audio and video media, the CSRC-list would often
   contain all participants who are not muted whereas text participants
   that don't type are completely silent and so don't show up in RTP
   packet CSRC-lists.


12.19.  Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00
        to -01

   Editorial cleanup.

   Changed capability indication from fmtp-parameter to SDP attribute

   Swapped order of redundancy elements in the example to match reality.

   Increased the SDP negotiation section

13.  References

13.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [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>.

   [RFC4102]  Jones, P., "Registration of the text/red MIME Sub-Type",
              RFC 4102, DOI 10.17487/RFC4102, June 2005,

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630,
              DOI 10.17487/RFC5630, October 2009,

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263,
              DOI 10.17487/RFC6263, June 2011,

   [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
              Thomson, "Session Traversal Utilities for NAT (STUN) Usage
              for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
              October 2015, <https://www.rfc-editor.org/info/rfc7675>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8825]  Alvestrand, H., "Overview: Real-Time Protocols for
              Browser-Based Applications", RFC 8825,
              DOI 10.17487/RFC8825, January 2021,

   [RFC8865]  Holmberg, C. and G. Hellström, "T.140 Real-Time Text
              Conversation over WebRTC Data Channels", RFC 8865,
              DOI 10.17487/RFC8865, January 2021,

   [RFC8866]  Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP:
              Session Description Protocol", RFC 8866,
              DOI 10.17487/RFC8866, January 2021,

   [T140]     ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for
              multimedia application text conversation", February 1998,

   [T140ad1]  ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000),
              Protocol for multimedia application text conversation",
              February 2000,

13.2.  Informative References

   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353,
              DOI 10.17487/RFC4353, February 2006,

   [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>.

   [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,

   [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,

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,

   [RFC8643]  Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T.
              Stach, "An Opportunistic Approach for Secure Real-time
              Transport Protocol (OSRTP)", RFC 8643,
              DOI 10.17487/RFC8643, August 2019,

Author's Address

   Gunnar Hellstrom
   Gunnar Hellstrom Accessible Communication
   SE-13670 Vendelso

   Email: gunnar.hellstrom@ghaccess.se