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Versions: (draft-hellstrom-avtcore-multi-party-rtt-source) 00 01 02 03 04 05 06 07 08 09 10 11 12

AVTCore                                                     G. Hellstrom
Internet-Draft                 Gunnar Hellstrom Accessible Communication
Updates: RFC 4103 (if approved)                         18 November 2020
Intended status: Standards Track
Expires: 22 May 2021


           RTP-mixer formatting of multi-party Real-time text
               draft-ietf-avtcore-multi-party-rtt-mix-10

Abstract

   Real-time text mixers for multi-party sessions need to identify the
   source of each transmitted group of text so that the text can be
   presented by endpoints in suitable grouping with other text from the
   same source, while new text from other sources is also presented in
   readable grouping as received interleaved in real-time.

   Regional regulatory requirements specify provision of real-time text
   in multi-party calls.  RFC 4103 mixer implementations can use
   traditional RTP functions for source identification, but the mixer
   source switching performance is limited when using the default
   transmission characteristics with redundancy.

   Enhancements for RFC 4103 real-time text mixing is provided in this
   document, suitable for a centralized conference model that enables
   source identification and source switching.  The intended use is for
   real-time text mixers and multi-party-aware participant endpoints.
   The specified mechanism build 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.

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

   The document updates RFC 4103[RFC4103]

   A specifications 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 22 May 2021.

Copyright Notice

   Copyright (c) 2020 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   license-info) in effect on the date of publication of this document.
   Please review these documents carefully, as they describe your rights
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   extracted from this document must include Simplified BSD License text
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   provided without warranty as described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  Selected solution and considered alternative  . . . . . .   5
     1.2.  Nomenclature  . . . . . . . . . . . . . . . . . . . . . .   7
     1.3.  Intended application  . . . . . . . . . . . . . . . . . .   8
   2.  Overview over the two specified solutions . . . . . . . . . .   8
     2.1.  Negotiated use of the RFC 4103 format for multi-party
           transmission in a single RTP stream . . . . . . . . . . .   9
     2.2.  Mixing for multi-party unaware endpoints  . . . . . . . .   9
   3.  Details for the multi-party aware mixing case . . . . . . . .   9
     3.1.  Offer/answer considerations . . . . . . . . . . . . . . .   9
     3.2.  Actions depending on capability negotiation result  . . .  10
     3.3.  Use of fields in the RTP packets  . . . . . . . . . . . .  10
     3.4.  Initial transmission of a BOM character . . . . . . . . .  11
     3.5.  Keep-alive  . . . . . . . . . . . . . . . . . . . . . . .  11
     3.6.  Transmission interval . . . . . . . . . . . . . . . . . .  11
     3.7.  Only one source per packet  . . . . . . . . . . . . . . .  11
     3.8.  Do not send received text to the originating source . . .  11
     3.9.  Clean incoming text . . . . . . . . . . . . . . . . . . .  11
     3.10. Redundancy  . . . . . . . . . . . . . . . . . . . . . . .  12
     3.11. Source switching  . . . . . . . . . . . . . . . . . . . .  12



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     3.12. Text placement in packets . . . . . . . . . . . . . . . .  12
     3.13. Empty T140blocks  . . . . . . . . . . . . . . . . . . . .  13
     3.14. Creation of the redundancy  . . . . . . . . . . . . . . .  13
     3.15. Timer offset fields . . . . . . . . . . . . . . . . . . .  13
     3.16. Other RTP header fields . . . . . . . . . . . . . . . . .  14
     3.17. Pause in transmission . . . . . . . . . . . . . . . . . .  14
     3.18. RTCP considerations . . . . . . . . . . . . . . . . . . .  14
     3.19. Reception of multi-party contents . . . . . . . . . . . .  15
     3.20. Performance considerations  . . . . . . . . . . . . . . .  17
     3.21. Security for session control and media  . . . . . . . . .  17
     3.22. SDP offer/answer examples . . . . . . . . . . . . . . . .  18
     3.23. Packet sequence example from a source switch  . . . . . .  19
     3.24. Maximum character rate "CPS"  . . . . . . . . . . . . . .  21
   4.  Presentation level considerations . . . . . . . . . . . . . .  21
     4.1.  Presentation by multi-party aware endpoints . . . . . . .  22
     4.2.  Multi-party mixing for multi-party unaware endpoints  . .  24
   5.  Relation to Conference Control  . . . . . . . . . . . . . . .  29
     5.1.  Use with SIP centralized conferencing framework . . . . .  30
     5.2.  Conference control  . . . . . . . . . . . . . . . . . . .  30
   6.  Gateway Considerations  . . . . . . . . . . . . . . . . . . .  30
     6.1.  Gateway considerations with Textphones (e.g.  TTYs).  . .  30
     6.2.  Gateway considerations with WebRTC. . . . . . . . . . . .  31
   7.  Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . .  31
   8.  Congestion considerations . . . . . . . . . . . . . . . . . .  31
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  32
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  32
     10.1.  Registration of the "rtt-mixer" sdp media attribute  . .  32
   11. Security Considerations . . . . . . . . . . . . . . . . . . .  33
   12. Change history  . . . . . . . . . . . . . . . . . . . . . . .  33
     12.1.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . .  33
     12.2.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . .  33
     12.3.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . .  34
     12.4.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . .  34
     12.5.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . .  34
     12.6.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . .  34
     12.7.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . .  34
     12.8.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . .  35
     12.9.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . .  36
     12.10. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . .  36



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     12.11. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-03 to
             draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . .  36
     12.12. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-02 to
             -03 . . . . . . . . . . . . . . . . . . . . . . . . . .  36
     12.13. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-01 to
             -02 . . . . . . . . . . . . . . . . . . . . . . . . . .  37
     12.14. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-00 to
             -01 . . . . . . . . . . . . . . . . . . . . . . . . . .  38
   13. References  . . . . . . . . . . . . . . . . . . . . . . . . .  38
     13.1.  Normative References . . . . . . . . . . . . . . . . . .  38
     13.2.  Informative References . . . . . . . . . . . . . . . . .  39
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  40

1.  Introduction

   RFC 4103[RFC4103] specifies use of RFC 3550 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.  RFC 4102 [RFC4102] registers the "text/red" format.
   Regional regulatory requirements specify provision of real-time text
   in multi-party calls.

   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 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 display."

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

   The redundancy scheme of RFC 4103 [RFC4103] enables efficient
   transmission of 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.




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   Real-time text mixers for multi-party sessions therefore need to
   insert the source of 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 while it is possible for a reading user
   to also 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 endpoint.

   There are existing implementations of RFC 4103 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 multi-party coded stream 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.  This 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 RFC 4103[RFC4103] by introducing an attribute
   for indicating capability for the multi-party mixing case and rules
   for source indications and source switching.

1.1.  Selected solution and considered alternative

   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.

   One RTP stream per source, sent in the same RTP session with
   "text/red" format.
      From some points of view, use of multiple RTP streams, one for
      each source, sent in the same RTP session, called the RTP
      translator model in [RFC3550], would be efficient, and use exactly
      the same packet format as [RFC4103], the same payload type and a
      simple SDP declaration.  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



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      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 not
      selected for inclusion in this document.

   The "text/red" format in RFC 4103 with shorter transmission
   interval, and indicating source in CSRC.
      The "text/red" format with "text/t140" payload in a single RTP
      stream can be sent with 100 ms packet intervals instead of the
      regular 300 ms.  The source is indicated in the CSRC field.
      Source switching can then be done every 100 ms while simultaneous
      transmission occurs.  With five participants sending text
      simultaneously, the switching and transmission performance is
      good.  With more simultaneously sending participants, there will
      be a noticable jerkiness in text presentation.  The jerkiness will
      be more expressed the more participants who send text
      simultaneously.  With ten sending participants, the jerkiness will
      be about one second.  Text sent from a source at the end of the
      period its text is sent by the mixer will have close to zero extra
      delay.  Recent text will be presented with no or low delay.  The
      one second jerkiness will be noticable 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, and in most applications also more than three
      parties in a call is 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.

   A new "text" media subtype with up to 15 sources in each packet.
      The mechanism makes use of the RTP mixer model specified in
      RFC3550[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.

   The presentation planned by the mixer for multi-party unaware
   endpoints.
      It is desirable to have a method that does not require any



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      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 has also 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
      [I-D.ietf-mmusic-t140-usage-data-channel].  That spcification
      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.

1.2.  Nomenclature

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

   The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP-
   mixer, RTP-translator are explained in [RFC3550]

   The term "T140block" is defined in RFC 4103 [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.

   "DTLS-SRTP" stands for security specified in RFC 5764 [RFC5764].







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   "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.

1.3.  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 want 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.

   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 over the two specified solutions

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




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2.1.  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.
   The method 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.  It is done by completing a negotiation
   for this kind of multi-party capability and by indicating source 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 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 for multi-party handling of real-time text.
   The solution 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 that acts on the presentation level.

   A party performing as a mixer, which has not negotiated the "rtt-
   mixer" sdp media attribute, but negotiated a "text/red" or "text/
   t140" format in a session with a participant SHOULD, 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.

3.  Details for the multi-party aware mixing case

3.1.  Offer/answer considerations

   RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a
   redundancy format "text/red" for increased robustness of real-time
   text transmission.  This document updates RFC 4103[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.

   The capability negotiation is based on use of the sdp media attribute
   "rtt-mixer".




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

   The syntax is as follows:
      "a=rtt-mixer"

3.2.  Actions depending on capability negotiation result

   A transmitting party SHALL send text according to the multi-party
   format only when the negotiation for this method was successful and
   when the CC field in the RTP packet is set to 1.  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 not performing as a mixer MUST not include the CSRC list.

3.3.  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 involved in a multi-
   party session, and otherwise 0.

   When transmitted from 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
   packet.  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].










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3.4.  Initial transmission of a BOM character

   As soon as a participant is known to participate in a session with
   another entity and being 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.

3.5.  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.
   Recommendations for keep-alive can be found in [RFC6263].

3.6.  Transmission interval

   A "text/red" transmitter in a mixer 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 330
   ms.  It is RECOMMENDED to send next 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 or equal to 100
   ms, and also the maximum character rate to the receiver is not
   exceeded.  The intention is to keep the latency low and network load
   limited while keeping a good protection against text loss in bursty
   packet loss conditions.

3.7.  Only one source per packet

   New and redundant 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.

3.8.  Do not send received text to the originating source

   Text received to a mixer from a participant SHOULD NOT be included in
   transmission from the mixer to that participant.

3.9.  Clean incoming text

   A mixer SHALL handle reception, recovery of 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.



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3.10.  Redundancy

   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
   SHOULD be set to the minimum of the number declared by the two
   parties negotiating a connection.  It is RECOMMENDED to transmit one
   original and two redundant generations of the T140blocks.

3.11.  Source switching

   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 received text in the mixer SHOULD be next
   in turn to get all its available unsent text transmitted.

3.12.  Text placement in packets

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

   *  100 ms has passed since the latest transmission to that receiver,
      and there is unsent text available for transmission.

   *  New text has arrived and more than 100 ms has passed since latest
      transmission to that receiver.

   *  330 ms has passed since already transmitted text was queued for
      transmission as redundant text, and more than 100 ms has passed
      since the latest transmission to that receiver, and the redundant
      text is still not sent.

   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.

   The SSRC of the source shall be placed as the only member in the
   CSRC-list.






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   Note: The CSRC-list in an RTP packet only includes the participant
   who's 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.

3.13.  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.

3.14.  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
   transmitter.

   The T140blocks saved for transmission as redundant data are assigned
   a planned transmission time 330 ms after the current time.

3.15.  Timer offset fields

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





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   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.

3.16.  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].

3.17.  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 can stop until either
   new T140blocks arrive, or a keep-alive method calls for transmission
   of keep-alive packets.

3.18.  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
   source.

   Integrity considerations SHALL be considered when composing these
   fields.












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3.19.  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.

3.19.1.  Multi-party vs two-party use

   If the "CC" field value of a received packet is 1, it indicates that
   multi-party transmission is active, and the receiver MUST be prepared
   to act on the source according to its role.  If the CC value is 0,
   the connection is point-to-point.

3.19.2.  Level of redundancy

   The used level of redundancy generations SHALL be evaluated from the
   received packet contents.  The number of generations (including the
   primary) is equal to the number of members in the redundancy header.

3.19.3.  Empty T140blocks

   Empty T140blocks are included as fillers for unused primary or
   redundancy levels in the packets.  They just do not provide any
   contents and do not contribute to the received streams.

3.19.4.  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 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 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 lost packets until any text may be lost, but text loss can



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   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.  Then a t140block
   SHALL 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 SHOULD
   rather falsely mark possible loss when there was no loss instead of
   not marking possible loss when there was loss.

3.19.5.  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.

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

   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 that 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.











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   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.

3.19.6.  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.

3.20.  Performance considerations

   This solution has good performance for up to five participants
   simultaneously sending text.  At higher numbers of participants
   simultaneously sending text, a jerkiness is visible in the
   presentation of text.  With ten participants simultaneously
   transmitting text, the jerkiness is about one second.  Even so, the
   transmission of text catches up, so there is no resulting total delay
   introduced.  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 long switching times.  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.

3.21.  Security for session control and media

   Security SHOULD be applied on both session control and media.  In
   applications where legacy endpoints without security may exist, a
   negotiation SHOULD be performed to decide if security by encryption
   will be applied.  If no other security solution is mandated for the
   application, then RFC 8643 OSRTP[RFC8643] SHOULD 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 security is
   shown in a couple of the following examples.







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3.22.  SDP offer/answer examples

   This sections 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.

     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
           a=rtt-mixer

      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
           a=rtt-mixer

      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
            a=rtt-mixer

   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
            a=rtt-mixer

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





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     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

3.23.  Packet sequence example from a source switch

   This example shows a symbolic flow of packets from a mixer including
   loss and recovery.  The sequence includes a source switch.  A and B
   are sources of RTT.  P indicates primary data.  R1 is first redundant
   generation data and R2 is second redundant generation data.  A1, B1,
   A2 etc are text chunks (T140blocks) received from the respective
   sources.  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      |
     |CC=1            |
     |CSRC list A     |
     |R2: A1          |
     |R1: A2          |
     |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 and will send that text 100 ms after packet 101.
   Transmission of A2 and A3 as redundancy is planned for 200 ms after
   packet 101.

     |----------------|
     |Seq no 102      |
     |CC=1            |
     |CSRC list B     |
     |R2  Empty       |
     |R1: Empty       |
     |P:  B1          |
     |----------------|
     B1 is retrieved from this packet. Redundant transmission of
     B1 is planned 200 ms after packet 102.






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     X----------------|
     X Seq no 103     |
     X CC=1           |
     X CSRC list A    |
     X R2: A2         |
     X R1: A3         |
     X P:  Empty      |
     X----------------|
     Packet 103 is assumed to be dropped in network problems. It
     contains redundancy for A. Sending A3 as second level
     redundancy is planned for 100 ms after packet 104.

     X----------------|
     X Seq no 104     |
     X CC=1           |
     X CSRC list B    |
     X R2: Empty      |
     X R1: B1         |
     X P2: B2         |
     X----------------|
     Packet 104 contains text from B, assumed dropped in network
     problems. The mixer has A3 redundancy to send and plans it
     after 100 ms.

     X----------------|
     X Seq no 105     |
     X CC=1           |
     X CSRC list A    |
     X R2: A3         |
     X R1: Empty      |
     X P:  Empty      |
     X----------------|
     Packet 105 is assumed to be dropped in network problems
     B1 and B2 still needs to be transmitted as redundancy.
     This is planned 320 ms after packet 105.

     |----------------|
     |Seq no 106      |
     |CC=1            |
     |CSRC list B     |
     | R2: B1         |
     | R1: B2         |
     | P:  Empty      |
     |----------------|







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   Packet 106 is received 320 ms after packet 105.  The latest received
   sequence number was 102.  103,104,105 were lost.  Three packets were
   thus lost during less than one second.  The simple rule for detection
   of text loss then results in that an indicator for possible loss
   (U'FFFD) [T140ad1], should be inserted generally for the mixer.

   The second level redundancy in packet 106 is B1 and has timestamp
   offset 620 ms.  The timestamp of packet 106 minus 620 is the
   timestamp of packet 101 which was received.  So B1 does not need to
   be retrieved.  The first level redundancy in packet 106 is 420.  The
   timestamp of packet 106 minus 420 is later than the latest received
   packet with source A.  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.

3.24.  Maximum character rate "CPS"

   The default maximum rate of reception of "text/t140" real-time text
   is in RFC 4103 [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 is the default for the "text/t140" stream in the
   "text/red" format when multi-party real-time text is negotiated.  See
   RFC 4103 [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

   ITU-T T.140 [T140] provides the presentation level requirements for
   the RFC 4103 [RFC4103] transport.  T.140 [T140] has functions for
   erasure and other formatting functions and 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."





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   Strict application of T.140 [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.

   T.140 [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.  Line
   Separator 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
   pause.

   When text is received from multiple original sources, the
   presentation SHOULD 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.

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




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

   Figure 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   |
   |___________________________________________________________________|



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   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 multi-party capability by the "rtt-
   mixer" sdp attribute 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.

   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 ITU-T
   T.140 [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 SHOULD be composed and sent for the source
   initially in the session and after each source switch.  With this
   procedure the time for source switching 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] SHOULD 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.







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4.2.2.  Actions by the mixer for transmission to the recipients

   The following procedure SHOULD be applied for each recipient of
   multi-part text from the mixer.

   The text for transmission SHOULD 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.  When there is text available for
   transmission from the mixer to a receiving party from more than one
   participant, the mixer SHOULD switch between transmission of text
   from the different sources at suitable points in the transmitted
   stream.

   When switching source, the mixer SHOULD insert a line separator if
   the already transmitted text did not end with a new line (line
   separator or CRLF).  A label SHOULD 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 SHOULD be delimited by suitable characters (e.g. '[
   ]') and transmitted.  The CSRC SHOULD indicate the selected source.
   Then text from that selected participant SHOULD 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 SHOULD 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
      started.





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

   The status SHOULD 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 SHOULD 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 SHOULD 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 SHOULD 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 SHOULD 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 SHOULD not be altered.

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

   SOS 0098  Start of string, used as a general protocol element



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      introducer, followed by a maximum 256 bytes string and the ST.
      The display count SHOULD not be altered.

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

   ESC 001B  Escape - used in control strings.  The display count SHOULD
      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.  SHOULD be deleted from
      incoming streams.  Shall 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.
      SHOULD be inserted by the reception procedure in case of
      unrecoverable loss of packets.  The display count SHOULD 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 is decreased by one when the
      BS is sent.  If the display count is at zero, it is not altered.

4.2.5.  Packet transmission

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



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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 makes 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.














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     _________________________________________________
    |       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.

     _________________________________________________
    |                                              |^|
    |[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            | |
    |                                              |-|
    |______________________________________________|v|
    | 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.

5.  Relation to Conference Control




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5.1.  Use with SIP centralized conferencing framework

   The SIP conferencing framework, mainly specified in RFC
   4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[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 users.

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 [RFC4566].

   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
   (TTY).

   More information on gateways to textphones (TTYs) is found in RFC
   5194[RFC5194]








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6.2.  Gateway considerations with WebRTC.

   Gateway operation to real-time text in WebRTC may also be required.
   In WebRTC, RTT is specified in
   [I-D.ietf-mmusic-t140-usage-data-channel].

   A multi-party bridge may have functionality for communicating by RTT
   both in RTP streams with RTT and WebRTC t140 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 t140 data channel.

   In WebRTC, it is assumed that for a multi-party session, one t140
   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 t140 channel SHOULD be established to WebRTC users with
   the label parameter composed from the NAME field in RTCP on the RTP
   side.

   When a new participant has entered the multi-party session with RTT
   transport in a WebRTC t140 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.

7.  Updates to RFC 4103

   This document updates RFC 4103[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 RFC 4103
   [RFC4103] are valid also in multi-party situations.

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



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   If the unlikely situation appears that more than 20 participants in a
   conference send text simultaneously, it will take more than 7 seconds
   between presentation of text from each of these participants.  More
   time than that can cause confusion in the session.  It is therefore
   RECOMMENDED that the mixer discards such text in excess 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.

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 3.1

   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
      member.

   Charset Dependent:  no

   O/A procedure:  See RFCXXXX Section 3.1

   Mux Category:  normal

   Reference:  RFCXXXX




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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 of
   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 masquerading
   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
   participants.

12.  Change history

12.1.  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
   method.

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

12.2.  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
   example.

   A number of editorial corrections and improvements.




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12.3.  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.

12.4.  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..

12.5.  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
   requirements.

   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.

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

   nomenclature and editorial improvements

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

12.7.  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.





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12.8.  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.
   -done-

   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-





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

12.9.  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.

12.10.  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
   inserted.

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

   9.  Gateway to WebRTC started.  More needed.

12.11.  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.

12.12.  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
   etc.



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   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.

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

   In Abstract and 1.  Introduction: Introduced wording about regulatory
   requirements.

   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
   added.

   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
   transport."

   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.





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

   Editorial cleanup.

   Changed capability indication from fmtp-parameter to SDP attribute
   "rtt-mix".

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

   Increased the SDP negotiation section

13.  References

13.1.  Normative References

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

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

   [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,
              <https://www.rfc-editor.org/info/rfc4102>.

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

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <https://www.rfc-editor.org/info/rfc4566>.







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   [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,
              <https://www.rfc-editor.org/info/rfc5764>.

   [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,
              <https://www.rfc-editor.org/info/rfc6263>.

   [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,
              <https://www.rfc-editor.org/info/rfc8643>.

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

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

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,
              <https://www.rfc-editor.org/info/rfc4353>.

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

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

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



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Author's Address

   Gunnar Hellstrom
   Gunnar Hellstrom Accessible Communication
   SE-13670 Vendelso
   Sweden

   Email: gunnar.hellstrom@ghaccess.se











































Hellstrom                  Expires 22 May 2021                 [Page 40]


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