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

AVTCore                                                     G. Hellstrom
Internet-Draft                 Gunnar Hellstrom Accessible Communication
Updates: RFC 4103 (if approved)                           12 August 2020
Intended status: Standards Track
Expires: 13 February 2021


           RTP-mixer formatting of multi-party Real-time text
               draft-ietf-avtcore-multi-party-rtt-mix-08

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.

   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/red" format 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-mix-rtp-
   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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   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on 13 February 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
   Provisions Relating to IETF Documents (https://trustee.ietf.org/
   license-info) in effect on the date of publication of this document.
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.  Code Components
   extracted from this document must include Simplified BSD License text
   as described in Section 4.e of the Trust Legal Provisions and are
   provided without warranty as described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Selected solution and considered alternative  . . . . . .   4
     1.2.  Nomenclature  . . . . . . . . . . . . . . . . . . . . . .   6
     1.3.  Intended application  . . . . . . . . . . . . . . . . . .   7
   2.  Specified solutions . . . . . . . . . . . . . . . . . . . . .   7
     2.1.  Negotiated use of the RFC 4103 format for multi-party in a
           single RTP stream . . . . . . . . . . . . . . . . . . . .   7
     2.2.  Mixing for multi-party unaware endpoints  . . . . . . . .  19
   3.  Presentation level considerations . . . . . . . . . . . . . .  19
     3.1.  Presentation by multi-party aware endpoints . . . . . . .  20
     3.2.  Multi-party mixing for multi-party unaware endpoints  . .  22
   4.  Gateway Considerations  . . . . . . . . . . . . . . . . . . .  27
     4.1.  Gateway considerations with Textphones (e.g.  TTYs).  . .  28
     4.2.  Gateway considerations with WebRTC. . . . . . . . . . . .  28
   5.  Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . .  29
   6.  Congestion considerations . . . . . . . . . . . . . . . . . .  29
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  29
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  29
     8.1.  Registration of the "rtt-mix-rtp-mixer" sdp media
           attribute . . . . . . . . . . . . . . . . . . . . . . . .  29



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   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  30
   10. Change history  . . . . . . . . . . . . . . . . . . . . . . .  30
     10.1.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . .  30
     10.2.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . .  31
     10.3.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . .  31
     10.4.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . .  31
     10.5.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . .  31
     10.6.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . .  31
     10.7.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . .  32
     10.8.  Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . .  33
     10.9.  Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-03 to
             draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . .  33
     10.10. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-02 to
             -03 . . . . . . . . . . . . . . . . . . . . . . . . . .  33
     10.11. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-01 to
             -02 . . . . . . . . . . . . . . . . . . . . . . . . . .  34
     10.12. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-00 to
             -01 . . . . . . . . . . . . . . . . . . . . . . . . . .  34
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  35
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  35
     11.2.  Informative References . . . . . . . . . . . . . . . . .  36
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  36

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.






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   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.  An assumption has to be made that the
   redundant parts in a packet are 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 is 300 ms.

   A mixer, selecting between text input from different sources and
   transmitting it in a common stream needs to make sure that the
   receiver can assign the received text to the proper sources for
   presentation.  Therefore, using RFC 4103 without any extra rule for
   source identification, the mixer needs to stop sending new text from
   one source and then make sure that all text sent so far has been sent
   with all intended redundancy levels (usually two) before switching to
   another source.  That causes the long time of one second to switch
   between transmission of text from one source to text from another
   source when using the default transmission interval 300 ms.  Both the
   total throughput and the switching performance in the mixer would be
   too low for most applications.  However by shorting the transmission
   interval to 100 ms, good performance is achieved for up to 3
   simultaneously sending sources and usable performance for up to 5
   simultaneously sending sources.  Capability to use this method is
   indicated by an sdp media attribute "rtt-mix-rtp-mixer".

   A negotiation mechanism can therefore be based on selection of the
   "text/red" with media attribute "rtt-mix-rtp-mixer" for verification
   that the parties are able to handle a multi-party coded stream and
   agreeing on using that method.

   A fall-back mixing procedure is specified for cases when the
   negotiation results in "text/red" without the "rtt-mix-rtp-mixer"
   attribute being the only common format for real-time text.

   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



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      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, there is currently lack of
      support for multi-stream RTP in certain implementation
      technologies.  This fact made it not included in this
      specification.

   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 300 ms while simultaneous
      transmission occurs.  With two participants sending text
      simultaneously, the switching and transmission performance is
      good.  With three 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 five sending participants, the jerkiness
      will be about 1400 ms.  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
      1400 ms 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 rare, and in most applications also more than three parties in
      a call is rare, this method can be used successfully without its
      limitations becoming annoying.  Negotiation is based on a new sdp
      media attribute "rtt-mix-rtp-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 specification.






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   The presentation planned by the mixer for multi-party unaware
   endpoints.
      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 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
      specification.

   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 specification is RTP transport.
      Therefore, even if the WebRTC transport provides good multi-party
      performance, it is just mentioned in this specification 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" stnds for security specified in RFC 5764 [RFC5764].



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1.3.  Intended application

   The method for multi-party real-time text documented in this
   specification 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.  Specified solutions

2.1.  Negotiated use of the RFC 4103 format for multi-party in a single
      RTP stream

   This section specifies use of the current format specified in
   [RFC4103] for true multi-party real-time text.  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.

   Please use [RFC4103] as reference when reading the following
   description.




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2.1.1.  Negotiation for use of this method

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

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

   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.

2.1.2.  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 is 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 packts are equal to what
   is specified in [RFC4103].

2.1.3.  Transmission of multi-party contents

   As soon as a participant is known to participate in a session and
   being available for text reception, a Unicode BOM character SHALL be
   sent to it 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.









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2.1.4.  Keep-alive

   After that, the transmitter SHALL send keep-alive traffic to the
   receivers 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].

2.1.5.  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 300
   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 while keeping a
   good protection against text loss in bursty packet loss conditions.

2.1.6.  Only one source per packet

   New and redundant text from one source MAY be transmitted in the same
   packet.  Text from different sources MUST NOT be transmitted in the
   same packet.

2.1.7.  Do not send received text to the originating source

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

2.1.8.  Clean incoming text

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

2.1.9.  Redundancy

   The transmitting party using redundancy SHALL send redundant
   repetitions of T140blocks aleady transmitted in earlier packets.

   The number of redundant generations of T140blocks to include in
   transmitted packets SHALL be deducted from the SDP negotiation.  It
   SHOULD be set to the minimum of the number declared by the two
   parties negotiating a connection.





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2.1.10.  Text placement in packets

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

   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 once their text
   have been transmitted as primary and the intended number of redundant
   generations.

2.1.11.  Source switching

   When text from more than one source is available for transmission,
   the mixer SHALL let the sources take turns in having their text
   transmitted.  When switching from transmission of one source to allow
   another source to have its text sent, all intended redundant
   generations of the last text from the current source MUST be
   transmitted before text from another source can be transmitted.

   Actively transmitting sources SHOULD be allowed to take turns as
   frequently as possible to have their text transmitted.  That implies
   that with the recommended redundancy, the mixer SHALL send primary
   text and two packets with redundant text from the current source
   before text from another source is transmitted.  The source with the
   oldest text received in the mixer SHOULD be next in turn to get all
   its available text transmitted.

2.1.12.  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.







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2.1.13.  Creation of the redundancy

   The primary T140block from a source in the latest transmitted packet
   is used to populate the first redundant T140block for that source.
   The first redundant T140block for that source from the latest
   transmission is placed as the second redundant T140block.

   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.

2.1.14.  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 from its original source 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.

2.1.15.  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.

   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].

2.1.16.  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.






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2.1.17.  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.

2.1.18.  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.

2.1.18.1.  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.

2.1.18.2.  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.

2.1.18.3.  Extracting text and handling recovery and loss

   The RTP sequence numbers of the received packets SHALL be monitored
   for gaps and packets out of order.

   As long as the sequence is correct, each packet SHALL be unpacked in
   order.  The T140blocks SHALL be extracted from the primary area, and
   the corresponding SSRC SHALL be extracted from the CSRC list and used
   for assigning the new T140block to the correct presentation areas (or
   correspondingly for other receiver roles).

   If a sequence number gap appears and is still there after some
   defined time for jitter resolution, T140data SHALL be recovered from
   redundant data.  If the gap is wider than the number of generations
   of redundant T140blocks in the packet, then a t140block SHALL be



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   created with a marker for possible text loss [T140ad1] and assigned
   to the SSRC of the transmitter as a general input from the mixer
   because in general it is not possible to deduct from which source(s)
   text was lost.  It is in some cases possible to deduct that no text
   was lost even for a gap wider than the redundancy generations, and in
   some cases it can be concluded which source that likely had loss.
   Therefore, the receiver MAY insert the marker for possible text loss
   [T140ad1] in the presentation area corresponding to the source which
   may have had loss.

   Then, the T140block in the received packet SHALL be retrieved
   beginning with the highest redundant generation, and assigning it to
   the presentation area of that source.  Finally the primary T140block
   SHALL be retrieved from the packet and similarly assigned to the
   corresponding presentation area for the source.

   If the sequence number gap was equal to or less than the number of
   redundancy generations in the received packet, a missing text marker
   SHALL NOT be inserted, and instead the T140block and the SSRC fully
   recovered from the redundancy information and the CSRC-list in the
   way indicated above.

2.1.18.4.  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 on reception.

2.1.18.5.  Empty T140blocks

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

2.1.19.  Performance considerations

   This solution has good performance for up to three participants
   simultaneously sending text.  At higher numbers of participants
   simultaneously sending text, a jerkiness is visible in the
   presentation of text.  With five participants simultaneously
   transmitting text, the jerkiness is about 1400 ms.  Evenso, 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



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   only the number of users sending text within the same moment that
   causes jerkiness, not the total number of users with RTT capability.

2.1.20.  Offer/answer considerations

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

   A party which has negotiated the the "rtt-mix-rtp-mixer" sdp media
   attribute MUST interpret the contents of the "CC" field the CSRC-list
   and the packets according to this section in received rtp packets in
   the corresponding RTP stream.

   A party performing as a mixer, which has not negotiated the "rtt-mix-
   rtp-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 Section 3.2.

   A party not performing as a mixer MUST not include the CSRC list.

2.1.21.  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 between security and no security SHOULD be applied.  If
   no other security solution is mandated by 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.

2.1.22.  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.









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     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-mix-rtp-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-mix-rtp-mixer

      Offer example for "text/red" format including multi-party
      and security:
            a=fingerprint: SHA-1 \
            4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
            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-mix-rtp-mixer

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

       Answer example from a multi-party capable device with security
            a=fingerprint: SHA-1 \
            FF:FF:FF:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
            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-mix-rtp-mixer

   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





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2.1.23.  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.

     |----------------|
     |Seq no 1        |
     |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 1 and assigned to reception
   area A.  The mixer is now assumed to have received text from source B
   and need to prepare for sending that text.  First it must send the
   redundant generations of text A2 and A3.

     |----------------|
     |Seq no 2        |
     |CC=1            |
     |CSRC list A     |
     |R2  A2          |
     |R1: A3          |
     |P: Empty        |
     |----------------|
     Nothing needs to be retrieved from this packet.

     X----------------|
     X Seq no 3       |
     X CC=1           |
     X CSRC list A    |
     X R2: A3         |
     X R1: Empty      |
     X P:  Empty      |
     X----------------|
     Packet 3 is assumed to be dropped in network problems. It was the
     last packet with contents from A before the source switch.






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     X----------------|
     X Seq no 4       |
     X CC=1           |
     X CSRC list B    |
     X R2: Empty      |
     X R1: Empty      |
     X P2: B1         |
     X----------------|
     Packet 4 contains text from B, assumed dropped in network problems.
     The mixer is assumed to have received text from A on turn to send.
     Sending of text from B must therefore be temporarily ended by
     sending redundancy twice.

     X----------------|
     X Seq no 5       |
     X CC=1           |
     X CSRC list B    |
     X R2: Empty      |
     X R1: B1         |
     X P:  Empty      |
     X----------------|
     Packet 5 is assumed to be dropped in network problems

     |----------------|
     |Seq no 6        |
     |CC=1            |
     |CSRC list B     |
     | R2: B1         |
     | R1: Empty      |
     | P:  Empty      |
     |----------------|

   Packet 6 is received.  The latest received sequence number was 2.
   Recovery is therefore tried for 3,4,5.  There is no coverage for seq
   no 3.  But knowing that A1 must have been sent as R2 in packet 3, it
   can be concluded that nothing was lost.

   For seqno 4, text B1 is recovered from the second generation
   redundancy and appended to the reception area of B.  For seqno 5,
   nothing needs to be recovered.  No primary text is available in
   packet 6.

   After this sequence, A3 and B1 have been received.  In this case no
   text was lost.  Even if also packet 2 was lost, it can be concluded
   that no text was lost.






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   If also packets 1 and 2 were lost, there would be a need to create a
   marker for possibly lost text (U'FFFD) [T140ad1], inserted generally
   and possibly also in text sequences A and B.

2.1.24.  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.

2.1.25.  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.

2.1.26.  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.










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2.2.  Mixing for multi-party unaware endpoints

   A method is specified in this section for cases when the
   participating endpoint does not implement any solution for multi-
   party presentation 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 and is further
   specified in Section 3.2.

3.  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."

   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.





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3.1.  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 simultaneously,
   the presentation SHOULD provide a view where text is added in
   multiple places simultaneously.

   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.

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

   When the mixer has indicated multi-party capability by the "rtt-mix-
   rtp-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.

3.2.1.  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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3.2.2.  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.

3.2.3.  Actions on transmission of text

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

3.2.4.  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.

3.2.5.  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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3.2.6.  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.

3.2.7.  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.

4.  Gateway Considerations




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4.1.  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]

4.2.  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.




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   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.

5.  Updates to RFC 4103

   This document updates RFC 4103[RFC4103] by introducing an sdp media
   attribute "rtt-mix-rtp-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.

6.  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.

   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.

7.  Acknowledgements

   James Hamlin for format and performance aspects.

8.  IANA Considerations

8.1.  Registration of the "rtt-mix-rtp-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-mix-rtp-mixer".

   Contact name:  IESG

   Contact email:  iesg@ietf.org

   Attribute name:  rtt-mix-rtp-mixer



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   Attribute syntax:  a=rtt-mix-rtp-mixer

   Attribute semantics:  See RFCXXXX Section 2.1.1

   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 2.1.20

   Mux Category:  normal

   Reference:  RFCXXXX

9.  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.

10.  Change history

10.1.  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.






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10.2.  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-mix-rtp-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..

10.3.  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.

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

   nomenclature and editorial improvements

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

10.5.  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.

10.6.  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--



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   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-

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

10.7.  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.



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10.8.  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.

10.9.  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.

10.10.  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.

   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.








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10.11.  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.


10.12.  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".



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

   Increased the SDP negotiation section

11.  References

11.1.  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>.

   [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>.



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   [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>.

11.2.  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>.

Author's Address

   Gunnar Hellstrom
   Gunnar Hellstrom Accessible Communication
   Esplanaden 30
   SE-13670 Vendelso
   Sweden

   Email: gunnar.hellstrom@ghaccess.se





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