AVTCore                                                     G. Hellstrom
Internet-Draft                 Gunnar Hellstrom Accessible Communication
Updates: 4103 (if approved)                                  25                                  26 May 2021
Intended status: Standards Track
Expires: 26 27 November 2021

           RTP-mixer formatting of multiparty Real-time text
               draft-ietf-avtcore-multi-party-rtt-mix-19
               draft-ietf-avtcore-multi-party-rtt-mix-20

Abstract

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

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

   A capability exchange is specified so that it can be verified that a
   mixer and a participant can handle the multiparty-coded real-time
   text stream using the RTP-mixer method.  The capability is indicated
   by use of an RFC 8866 Session Description Protocol (SDP) media
   attribute "rtt-mixer".

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

   A specification of how a mixer can format text for the case when the
   endpoint is not multiparty-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.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on 26 27 November 2021.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   6
     1.2.  Selected solution and considered alternatives . . . . . .   7
     1.3.  Intended application  . . . . . . . . . . . . . . . . . .   9
   2.  Overview of the two specified solutions and selection of
           method  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     2.1.  The RTP-mixer-based solution for multiparty-aware
           endpoints . . . . . . . . . . . . . . . . . . . . . . . .  10
     2.2.  Mixing for multiparty-unaware endpoints . . . . . . . . .  11
     2.3.  Offer/answer considerations . . . . . . . . . . . . . . .  11
     2.4.  Actions depending on capability negotiation result  . . .  13
   3.  Details for the RTP-mixer-based mixing method for
           multiparty-aware endpoints  . . . . . . . . . . . . . . .  13
     3.1.  Use of fields in the RTP packets  . . . . . . . . . . . .  13
     3.2.  Initial transmission of a BOM character . . . . . . . . .  14
     3.3.  Keep-alive  . . . . . . . . . . . . . . . . . . . . . . .  14
     3.4.  Transmission interval . . . . . . . . . . . . . . . . . .  14
     3.5.  Only one source per packet  . . . . . . . . . . . . . . .  15
     3.6.  Do not send received text to the originating source . . .  15
     3.7.  Clean incoming text . . . . . . . . . . . . . . . . . . .  16
     3.8.  Redundant transmission principles . . . . . . . . . . . .  16
     3.9.  Text placement in packets . . . . . . . . . . . . . . . .  16
     3.10. Empty T140blocks  . . . . . . . . . . . . . . . . . . . .  17
     3.11. Creation of the redundancy  . . . . . . . . . . . . . . .  17
     3.12. Timer offset fields . . . . . . . . . . . . . . . . . . .  18
     3.13. Other RTP header fields . . . . . . . . . . . . . . . . .  18
     3.14. Pause in transmission . . . . . . . . . . . . . . . . . .  18
     3.15. RTCP considerations . . . . . . . . . . . . . . . . . . .  19
     3.16. Reception of multiparty contents  . . . . . . . . . . . .  19
     3.17. Performance considerations  . . . . . . . . . . . . . . .  21
     3.18. Security for session control and media  . . . . . . . . .  21
     3.19. SDP offer/answer examples . . . . . . . . . . . . . . . .  22
     3.20. Packet sequence example from interleaved transmission . .  23
     3.21. Maximum character rate "cps"  . . . . . . . . . . . . . .  26
   4.  Presentation level considerations . . . . . . . . . . . . . .  26
     4.1.  Presentation by multiparty-aware endpoints  . . . . . . .  27
     4.2.  Multiparty mixing for multiparty-unaware endpoints  . . .  29
   5.  Relation to Conference Control  . . . . . . . . . . . . . . .  35
     5.1.  Use with SIP centralized conferencing framework . . . . .  36
     5.2.  Conference control  . . . . . . . . . . . . . . . . . . .  36
   6.  Gateway Considerations  . . . . . . . . . . . . . . . . . . .  36
     6.1.  Gateway considerations with Textphones  . . . . . . . . .  36
     6.2.  Gateway considerations with WebRTC  . . . . . . . . . . .  36
   7.  Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . .  37
   8.  Congestion considerations . . . . . . . . . . . . . . . . . .  38
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  38
   10.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  38
     10.1.
     9.1.  Registration of the "rtt-mixer" SDP media attribute . . .  38
   11.
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  39
   12.
   11. Change history  . . . . . . . . . . . . . . . . . . . . . . .  40
     12.1.
     11.1.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-20 . . . . . . .  40
     11.2.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-19 . . . . . . .  40
     12.2.
     11.3.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-18 . . . . . . .  40
     12.3.
     11.4.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-17 . . . . . . .  40
     12.4.
     11.5.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-16 . . . . . . .  40
     12.5.
     11.6.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . .  40
     12.6.  41
     11.7.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . .  41
     12.7.
     11.8.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . .  41
     12.8.
     11.9.  Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . .  41
     12.9.  42

     11.10. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . .  42

     12.10.
     11.11. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . .  42
     12.11.
     11.12. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . .  42
     12.12.
     11.13. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . .  43
     12.13.
     11.14. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . .  43
     12.14.
     11.15. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . .  43
     12.15.
     11.16. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . .  43
     12.16.
     11.17. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . .  43
     12.17.
     11.18. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . .  44
     12.18.
     11.19. Changes included in
             draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . .  45
     12.19.
     11.20. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . .  45
     12.20.
     11.21. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-03 to
             draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . .  45
     12.21.
     11.22. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-02 to
             -03 . . . . . . . . . . . . . . . . . . . . . . . . . .  45
     12.22.
     11.23. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-01 to
             -02 . . . . . . . . . . . . . . . . . . . . . . . . . .  46
     12.23.
     11.24. Changes from
             draft-hellstrom-avtcore-multi-party-rtt-source-00 to
             -01 . . . . . . . . . . . . . . . . . . . . . . . . . .  47
   13.
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  47
     13.1.
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  47
     13.2.
     12.2.  Informative References . . . . . . . . . . . . . . . . .  48
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  49
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  49

1.  Introduction

   "RTP Payload for Text Conversation" [RFC4103] specifies use of the
   Real-Time Transport Protocol (RTP) [RFC3550] for transmission of
   real-time text (RTT) and the "text/t140" format.  It also specifies a
   redundancy format "text/red" for increased robustness.  The "text/
   red" format is registered in [RFC4102].

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

   A requirement related to multiparty sessions from the presentation
   level standard T.140 [T140] for real-time text is: "The display of
   text from the members of the conversation should be arranged so that
   the text from each participant is clearly readable, and its source
   and the relative timing of entered text is visualized in the
   display."

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

   Use of RTT is increasing, and specifically, use in emergency calls is
   increasing.  Emergency call use requires multiparty mixing because it
   is common that one agent needs to transfer the call to another
   specialized agent but is obliged to stay on the call at least to
   verify that the transfer was successful.  Mixer implementations for
   RFC 4103 "RTP Payload for Text Conversation" can use traditional RFC
   3550 RTP functions for mixing and source identification, but the
   performance of the mixer when giving turns for the different sources
   to transmit is limited when using the default transmission
   characteristics with redundancy.

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

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

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

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

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

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

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

1.1.  Terminology

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

   The terms Source Description (SDES), Canonical name (CNAME), Name
   (NAME), Synchronization Source (SSRC), Contributing Source (CSRC),
   CSRC list, CSRC count [CC], Real-Time control protocol (RTCP), RTP-
   mixer, RTP-translator are defined in [RFC3550].

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

   "TTY" stands for a textphone type used in North America.

   Web based real-time communication (WebRTC) is specified by the World
   Wide Web Consortium (W3C) and IETF.  See [RFC8825].

   "DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension
   for use with Secure Real-Time Transport Protocol/Secure Real-Time
   Control Protocol (SRTP/SRTCP) specified in [RFC5764].

   "multiparty-aware" describes 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.

   "multiparty-unaware" describes an endpoint not itself being able to
   separate text from different sources when received through a common
   conference mixer.

1.2.  Selected solution and considered alternatives

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

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

   RTP-mixer-based method for multiparty-aware endpoints
      The "text/red" format in RFC 4103 is sent with a shorter
      transmission interval with the RTP-mixer method and indicating the
      source in the CSRC field.  The "text/red" format with a "text/
      t140" payload in a single RTP stream can be sent when text is
      available from the call participants instead of at the regular 300
      ms.  Transmission of packets with text from different sources can
      then be done smoothly while simultaneous transmission occurs as
      long as it is not limited by the maximum character rate "cps".
      With ten participants sending text simultaneously, the switching
      and transmission performance is good.  With more simultaneously
      sending participants, and with receivers having the default
      capacity there will be a noticeable jerkiness and delay in text
      presentation.  The jerkiness will be more expressed the more
      participants who send text simultaneously.  Two seconds jerkiness
      will be noticeable and slightly unpleasant, but it 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 expected to be
      very rare as described in Section 1.3, this method can be used
      successfully with good performance.  Recovery of text in case of
      packet loss is based on analysis of timestamps of received
      redundancy versus earlier received text.  Negotiation is based on
      a new SDP media attribute "rtt-mixer".  This method is selected to
      be the main one specified in this document.

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

   Mixing for multiparty-unaware endpoints
      Presentation of text from multiple parties is prepared by the
      mixer in one single stream.  It is desirable to have a method that
      does not require any modifications in existing user devices
      implementing RFC 4103 for RTT without explicit support of
      multiparty 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 vary.  The method also has
      other limitations, but is included in this document as a fallback
      method.  In calls where parties take turns properly by ending
      their entries with a new line, the limitations will have limited
      influence on the user experience. when only two parties send text,
      these two will see the text in real time with no delay.  This
      method is specified as a fallback method in this document.

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

1.3.  Intended application

   The method for multiparty real-time text specified in this document
   is primarily intended for use in transmission between mixers and
   endpoints in centralized 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 call
   taker wants to make an attended handover of a call to another agent,
   and stay to observe the session.  Multimedia conference sessions with
   support for participants to contribute in text is another
   application.  Conferences with central support for speech-to-text
   conversion is yet another mentioned application.

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

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

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

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

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

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

2.1.  The RTP-mixer-based solution for multiparty-aware endpoints

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

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

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

2.2.  Mixing for multiparty-unaware endpoints

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

   A mixer SHOULD by default format and transmit text to a call
   participant to be suitable to present on a multiparty-unaware
   endpoint which has not negotiated any method for true multiparty RTT
   handling, but negotiated a "text/red" or "text/t140" format in a
   session.  This SHOULD be done if nothing else is specified for the
   application in order to maintain interoperability.  Section 4.2
   specifies how this mixing is done.

2.3.  Offer/answer considerations

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

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

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

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

2.3.1.  Initial offer

   A party intending to set up a session and being willing to use the
   RTP-mixer-based method of this specification for sending or receiving
   or both sending and receiving real-time text SHALL include the "rtt-
   mixer" SDP attribute in the corresponding "text" media section in the
   initial offer.

   The party MAY indicate capability for both the RTP-mixer-based method
   of this specification and other methods.

   When the offeror has sent the offer including the "rtt-mixer"
   attribute, it MUST be prepared to receive and handle real-time text
   formatted according to both the method for multiparty-aware parties
   specified in Section 3 in this specification and two-party formatted
   real-time text.

2.3.2.  Answering the offer

   A party receiving an offer containing the "rtt-mixer" SDP attribute
   and being willing to use the RTP-mixer-based method of this
   specification for sending or receiving or both sending and receiving
   SHALL include the "rtt-mixer" SDP attribute in the corresponding
   "text" media section in the answer.

   If the offer did not contain the "rtt-mixer" attribute, the answer
   MUST NOT contain the "rtt-mixer" attribute.

   Even when the "rtt-mixer" attribute is successfully negotiated, the
   parties MAY send and receive two-party coded real-time text.

   An answer MUST NOT include acceptance of more than one method for
   multiparty real-time text in the same RTP session.

   When the answer including acceptance is transmitted, the answerer
   MUST be prepared to act on received text in the negotiated session
   according to the method for multiparty-aware parties specified in
   Section 3 of this specification.  Reception of text for a two-party
   session SHALL also be supported.

2.3.3.  Offeror processing the answer

   When the answer is processed by the offeror, it MUST act as specified
   in Section 2.4

2.3.4.  Modifying a session

   A session MAY be modified at any time by any party offering a
   modified SDP with or without the "rtt-mixer" SDP attribute expressing
   a desired change in the support of multiparty real-time text.

   If the modified offer adds indication of support for multiparty real-
   time text by including the "rtt-mixer" SDP attribute, the procedures
   specified in the previous subsections SHALL be applied.

   If the modified offer deletes indication of support for multiparty
   real-time text by excluding the "rtt-mixer" SDP attribute, the answer
   MUST NOT contain the "rtt-mixer" attribute.  After processing this
   SDP exchange, the parties MUST NOT send real-time text formatted for
   multiparty-aware parties according to this specification.

2.4.  Actions depending on capability negotiation result

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

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

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

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

3.  Details for the RTP-mixer-based mixing method for multiparty-aware
    endpoints

3.1.  Use of fields in the RTP packets

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

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

   When 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.

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

3.2.  Initial transmission of a BOM character

   As soon as a participant is known to participate in a session with
   another entity and is available for text reception, a Unicode Byte-
   Order Mark (BOM) character SHALL be sent to it by the other entity
   according to the procedures in this section.  This is useful in many
   configurations to open ports and firewalls and setting up the
   connection between the application and the network.  If the
   transmitter is a mixer, then the source of this character SHALL be
   indicated to be the mixer itself.

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

3.3.  Keep-alive

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

3.4.  Transmission interval

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

   The reason for the value 330 ms is that many sources of text will
   transmit new text with 300 ms intervals during periods of continuous
   user typing, and then reception in the mixer of such new text will
   cause a combined transmission of the new text and the unsent
   redundancy from the previous transmission.  Only when the user stops
   typing, the 330 ms interval will be applied to send the redundancy.

   If the Characters Per Second (cps) value is reached, a longer
   transmission interval SHALL be applied for text from all sources as
   specified in [RFC4103] and only as much of the text queued for
   transmission SHALL be sent at the end of each transmission interval
   as can be allowed without exceeding the "cps" value.  Division of
   text for partial transmission MUST then be made at T140block borders.
   When the transmission rate falls under the "cps" value again, the
   transmission intervals SHALL be returned to 330 ms and transmission
   of new text SHALL return to be made as soon as new text is available.

   NOTE: that extending the transmission intervals during high load
   periods does not change the number of characters to be conveyed.  It
   just evens out the load in time and reduces the number of packets per
   second.  With human created conversational text, the sending user
   will eventually take a pause letting transmission catch up.

   See also Section 8.

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

3.5.  Only one source per packet

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

3.6.  Do not send received text to the originating source

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

3.7.  Clean incoming text

   A mixer SHALL handle reception, recovery from packet loss, deletion
   of superfluous redundancy, marking of possible text loss and deletion
   of 'BOM' characters from each participant before queueing received
   text for transmission to receiving participants as specified in
   [RFC4103] for single-party sources and Section 3.16 for multiparty
   sources (chained mixers).

3.8.  Redundant transmission principles

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

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

3.9.  Text placement in packets

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

   *  The transmission interval is the normal 330 ms and there is newly
      received unsent text available for transmission to that receiver.

   *  The current transmission interval has passed and is longer than
      the normal 330 ms and there is newly received unsent text
      available for transmission to that receiver.

   *  The current transmission interval ( normally 330 ms) has passed
      since already transmitted text was queued for transmission as
      redundant text.

   The principles from [RFC4103] apply for populating the header, the
   redundancy header and the data in the packet with specifics specified
   here and in the following sections.

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

   Redundant text SHALL also be included, and the assessment of how much
   new text can be included within the maximum packet size MUST take
   into account that the redundancy has priority to be transmitted in
   its entirety.  See Section 3.4

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

3.10.  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 populated with the available
   T140blocks for the source in turn to be sent as primary, and empty
   T140blocks for the agreed number of redundancy generations.

3.11.  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 the 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 the next
   transmission of text from that source.

   Usually this is the level of redundancy used.  If a higher level 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, but SHOULD
   be transmitted earlier if new text for the same source gets in turn
   for transmission before that time.

3.12.  Timer offset fields

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

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

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

3.13.  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 self-sourced data.

   The current time SHALL be inserted in the timestamp.

   The SSRC header field SHALL contain the SSRC of the RTP session where
   the packet will be transmitted.

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

3.14.  Pause in transmission

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

3.15.  RTCP considerations

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

   Privacy considerations SHALL be taken when composing these fields.
   They contain name and address information that may be sensitive to
   transmit in its entirety, e.g., to unauthenticated participants.

3.16.  Reception of multiparty 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.  They require
   considerations if the stream shall just be forwarded, or distributed
   based on the different sources.

3.16.1.  Acting on the source of the packet contents

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

3.16.2.  Detection and indication of possible text loss

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

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

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

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

3.16.3.  Extracting text and handling recovery

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

   When a packet is received in an RTP session using the packetization
   for multiparty-aware endpoints, its T140blocks SHALL be extracted in
   the following way.

   The description is adapted to the default
   redundancy case using the original and two redundant generations.

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

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

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

   If the packet is not the first packet from a source, then if the
   second generation
   redundant data is available, its the process SHALL start with the oldest
   generation.  The timestamp of that redundant data 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 all younger generations of redundant data.  After that, the
   timestamp of the packet SHALL be compared with the timestamp of the
   latest retrieved data from the same source and if it is later, then
   the primary data SHALL be retrieved from the packet and appended to
   the receive buffer for the source.

3.16.4.  Delete 'BOM'

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

3.17.  Performance considerations

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

3.18.  Security for session control and media

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

   Further general security considerations are covered in Section 11. 10.

   End-to-end encryption would require further work and could be based
   on WebRTC as specified in Section 1.2 or on double encryption as
   specified in [RFC8723].

3.19.  SDP offer/answer examples

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

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

     Offer example for "text/red" format and multiparty 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 multiparty-aware 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 multiparty
      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 multiparty-aware 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.

     Answer example from a multiparty-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

3.20.  Packet sequence example from interleaved transmission

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

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

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

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

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

     X------------------------|
     X Seq no 104, Timer=20800|
     X CC=1                   |
     X CSRC list B            |
     X R2: Empty, Offset=600  |
     X R1: B1, Offset=300     |
     X P:  B2                 |
     X------------------------|
     Packet 104 contains text from B, including new B2 and
     redundant B1. It is assumed dropped due to network
     problems.
     The mixer has A3 redundancy to send, but no new text
     appears from A and therefore the redundancy is sent
     330 ms after the previous packet with text from A.

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

     R2 is checked. Its original time was 21060-660=20400.
     A packet with text from A was received with that
     timestamp, so nothing needs to be recovered.

     B1 and B2 still need to be transmitted as redundancy.
     This is planned 330 ms after packet 104. That
     would be at 21130.

     |-----------------------|
     |Seq no 106, Timer=21130|
     |CC=1                   |
     |CSRC list B            |
     | R2: B1, Offset=630    |
     | R1: B2, Offset=330    |
     | P:  Empty             |
     |-----------------------|

   Packet 106 is received.

   The second level redundancy in packet 106 is B1 and has timestamp
   offset 630 ms.  The timestamp of packet 106 minus 630 is 20500 which
   is the timestamp of packet 102 that was received.  So B1 does not
   need to be retrieved.  The first level redundancy in packet 106 has
   offset 330.  The timestamp of packet 106 minus 330 is 20800.  That is
   later than the latest received packet with source B.  Therefore B2 is
   retrieved and assigned to the input buffer for source B.  No primary
   is available in packet 106.

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

3.21.  Maximum character rate "cps"

   The default maximum rate of reception of "text/t140" real-time text
   is in [RFC4103] specified to be 30 characters per second.  The actual
   rate is calculated without regard to any redundant text transmission
   and is in the multiparty case evaluated for all sources contributing
   to transmission to a receiver.  The value MAY be modified in the
   "cps" parameter of the FMTP attribute in the media section for the
   "text/t140" media.  A mixer combining real-time text from a number of
   sources may occasionally have a higher combined flow of text coming
   from the sources.  Endpoints SHOULD therefore specify a suitable
   higher value for the "cps" parameter, corresponding to its real
   reception capability.  A value for "cps" of 90 SHALL be the default
   for the "text/t140" stream in the "text/red" format when multiparty
   real-time text is negotiated.  See [RFC4103] for the format and use
   of the "cps" parameter.  The same rules apply for the multiparty case
   except for the default value.

4.  Presentation level considerations

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

   "The display of text from the members of the conversation should be
   arranged so that the text from each participant is clearly readable,
   and its source and the relative timing of entered text is visualized
   in the display.  Mechanisms for looking back in the contents from the
   current session should be provided.  The text should be displayed as
   soon as it is received."
   Strict application of [T140] is of essence for the interoperability
   of real-time text implementations and to fulfill the intention that
   the session participants have the same information conveyed in the
   text contents of the conversation without necessarily having the
   exact same layout of the conversation.

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

   There is no strict "message" concept in real-time text.  The Unicode
   Line Separator character SHALL be used as a separator allowing a part
   of received text to be grouped in presentation.  The characters
   "CRLF" may be used by other implementations as a replacement for Line
   Separator.  The "CRLF" combination SHALL be erased by just one
   erasing action, the same as the Line Separator.  Presentation
   functions are allowed to group text for presentation in smaller
   groups than the line separators imply and present such groups with
   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 multiparty-aware endpoints

   A multiparty-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 SHALL provide a view where text is added in multiple
   presentation fields.

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

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

                 _________________________________________________
                |                                              |^|
                |[Alice] Hi, Alice here.                       |-|
                |                                              | |
                |[Bob] Bob as well.                            | |
                |                                              | |
                |[Eve] Hi, this is Eve, calling from Paris.    | |
                |      I thought you should be here.           | |
                |                                              | |
                |[Alice] I am coming on Thursday, my           | |
                |      performance is not until Friday morning.| |
                |                                              | |
                |[Bob] And I on Wednesday evening.             | |
                |                                              | |
                |[Alice] Can we meet on Thursday evening?      | |
                |                                              | |
                |[Eve] Yes, definitely. How about 7pm.         | |
                |     at the entrance of the restaurant        | |
                |     Le Lion Blanc?                           | |
                |[Eve] we can have dinner and then take a walk |-|
                |______________________________________________|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   |
   |___________________________________________________________________|

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

4.2.  Multiparty mixing for multiparty-unaware endpoints

   When the mixer has indicated RTT multiparty capability in an SDP
   negotiation, but the multiparty 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
   multiparty real-time text in one stream intended to be presented by
   an endpoint with no multiparty awareness, when that is desired in the
   actual implementation.  The following specifies a procedure which MAY
   be applied in that situation.

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

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

   The mixer mixing for multiparty-unaware endpoints SHALL compose a
   simulated, limited multiparty 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 line separator
   between them if the transmitted text did not already end with a new
   line (line separator or CRLF).  A presentable label SHALL be composed
   and sent for the source initially in the session and after each
   source switch.  With this procedure the time for switching from
   transmission of text from one source to transmission of text from
   another source depends on the actions of the users.  In order to
   expedite source switching, a user can, for example, end its turn with
   a new line.

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

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

4.2.2.  Actions by the mixer for transmission to the recipients

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

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

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

   Information available to the mixer for composing the label may
   contain sensitive personal information that SHOULD NOT be revealed in
   sessions not securely authenticated and confidentiality protected.
   Privacy considerations regarding how much personal information is
   included in the label SHOULD therefore be taken when composing the
   label.

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

   *  A completed phrase ended by comma

   *  A completed sentence

   *  A new line (line separator or CRLF)

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

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

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

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

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

4.2.3.  Actions on transmission of text

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

4.2.4.  Actions on transmission of control codes

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

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

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

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

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

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

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

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

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

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

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

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

4.2.5.  Packet transmission

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

4.2.6.  Functional limitations

   When a multiparty-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 multiparty-unaware endpoints, there will be two levels
   of source indicators for the received text; one generated by the
   mixer and inserted in a label after each source switch, and another
   generated by the receiving endpoint and inserted after each switch
   between local and remote source in the presentation area.  This will
   waste display space and look inconsistent to the reader.

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

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

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

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

4.2.7.  Example views of presentation on multiparty-unaware endpoints

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

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

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

   This figure shows how a coordinated column view MAY be presented on
   Alice's device in a view with two-columns.  The mixer inserts labels
   to show how the sources alternate in the column with received text.

   The mixer alternates between the sources at suitable points in the
   text exchange so that text entries from each party can be
   conveniently read.

     _________________________________________________
    |                                              |^|
    |(Alice) Hi, Alice here.                       |-|
    |                                              | |
    |(mix)[Bob)] Bob as well.                      | |
    |                                              | |
    |[Eve] Hi, this is Eve, calling from Paris     | |
    |      I thought you should be here.           | |
    |                                              | |
    |(Alice) I am coming on Thursday, my           | |
    |      performance is not until Friday morning.| |
    |                                              | |
    |(mix)[Bob] And I on Wednesday evening.        | |
    |                                              | |
    |[Eve] we can have dinner and then walk        | |
    |                                              | |
    |[Eve] But I need to be back to                | |
    |    the hotel by 11 because I need            | |
    |                                              |-|
    |______________________________________________|v|
    | of course, I underst                           |
    |________________________________________________|

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

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

5.  Relation to Conference Control
5.1.  Use with SIP centralized conferencing framework

   The Session Initiation Protocol (SIP) conferencing framework, mainly
   specified in [RFC4353], [RFC4579] and [RFC4575] is suitable for
   coordinating sessions including multiparty 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 [RFC8866].

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

6.  Gateway Considerations

6.1.  Gateway considerations with Textphones

   multiparty RTT sessions may involve gateways of different kinds.
   Gateways involved in setting up sessions SHALL correctly reflect the
   multiparty 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 Public Switched Telephone
   Network (PSTN) for communication with textphones (e.g., TTYs).
   Textphones are limited devices with no multiparty awareness, and it
   SHOULD therefore be suitable for the gateway to not indicate
   multiparty awareness for that case.  Another solution is that the
   gateway indicates multiparty capability towards the mixer, and
   includes the multiparty mixer function for multiparty-unaware
   endpoints itself.  This solution makes it possible to adapt to the
   functional limitations of the textphone.

   More information on gateways to textphones is found in [RFC5194]

6.2.  Gateway considerations with WebRTC

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

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

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

   The T.140 data 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 T.140 data channel is used as the NAME field
   in RTCP to participants on the RTP side.  The other T.140 data
   channels are only for text from other participants to the WebRTC
   user.

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

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

   When a participant on the RTP side is disconnected from the
   multiparty session, the corresponding T.140 data channel(s) SHOULD be
   closed.

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

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

7.  Updates to RFC 4103

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

8.  Congestion considerations

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

   The time values SHALL then be applied per source of text sent to a
   receiver.

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

9.  Acknowledgements

   James Hamlin for format and performance aspects.

10.  IANA Considerations

10.1.

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

   [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
   RFC number of this document.]

   IANA is asked to register the new SDP attribute "rtt-mixer".

   Contact name:  IESG

   Contact email:  iesg@ietf.org

   Attribute name:  rtt-mixer

   Attribute semantics:  See RFCXXXX Section 2.3

   Attribute value:  none

   Usage level:  media

   Purpose:  Indicate support by mixer and endpoint of multiparty 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 2.3

   Mux Category:  normal

   Reference:  RFCXXXX

11.

10.  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 to maintain confidentiality
   and integrity of the RTT data.  This situation is similar to the
   situation for handling audio and video media in centralized mixers.

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

   The services available through the RTT mixer may have special
   interest for deaf and hard-of-hearing persons.  Some users may want
   to refrain from revealing such characteristics broadly in
   conferences.  The design of the conference systems where the mixer is
   included MAY need to be made with confidentiality of such
   characteristics in mind.

   Participants with malicious intentions may appear and e.g., disturb
   the multiparty session by emitting a continuous flow of text.  They
   may also send text that appears to originate from other participants.
   Counteractions should be to require secure signaling, media and
   authentication, and to provide higher-layer conference functions
   e.g., for blocking, muting, and expelling participants.

   Participants with malicious intentions may also try to disturb the
   presentation by sending incomplete or malformed control codes.
   Handling of text from the different sources by the receivers MUST
   therefore be well separated so that the effects of such actions only
   affect text from the source causing the action.

   Care should be taken that if use of the mixer is allowed for users
   both with and without security procedures, opens for possible attacks
   by both unauthenticated call participants and even eavesdropping and
   manipulating of content non-participants.

   As already stated in Section 3.18, security in media SHOULD be
   applied by using DTLS-SRTP [RFC5764] on the media level.

   Further security considerations specific for this application are
   specified in Section 3.18.

12.

11.  Change history

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

12.1.

11.1.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-20

   Inclusion of edits as respone to a comment by Benjamin Kaduk in
   section 3.16.3 to make the recovery procedure generic.

   Added persons to the acknowledgements and moved acknowledgements to
   last in the document.

11.2.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-19

   Edits because of comments in a review by Francesca Palombini.

   Edits because of comments from Benjamin Kaduk.

   Proposed to not change anything because of Robert Wilton's comments.

   Two added sentences in the security section to meet comments by Roman
   Danyliw.

12.2.

11.3.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-18

   Edits of nits as proposed in a review by Lars Eggert.

   Edits as response to review by Martin Duke.

12.3.

11.4.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17

   Actions on Gen-ART review comments.

   Actions on SecDir review comments.

12.4.

11.5.  Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16

   Improvements in the offer/answer considerations section by adding
   subsections for each phase in the negotiation as requested by IANA
   expert review.

12.5.

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

   Actions on review comments from Jurgen Schonwalder:

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

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

   Presentation examples with suource labels made less confusing, and
   explained.

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

   Reference to RFC 8825 inserted to explain WebRTC

   Nit in wording in terminology section adjusted.

12.6.

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

   Changes from comments by Murray Kucherawy during AD review.

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

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

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

12.7.

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

   Changed year to 2021.

   Changed reference to draft on RTT in WebRTC to recently published RFC
   8865.

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

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

12.8.

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

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

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

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

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

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

12.9.

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

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

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

12.10.

11.11.  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

12.11.

11.12.  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.

12.12.

11.13.  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.

12.13.

11.14.  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.

12.14.

11.15.  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.

12.15.

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

   nomenclature and editorial improvements

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

12.16.

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

   'Redundancy header' renamed to 'data header'.

   More clarifications added.

   Language and figure number corrections.

12.17.

11.18.  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-

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

12.18.

11.19.  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.

12.19.

11.20.  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 multiparty-unaware mixing method
   inserted.

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

   9.  Gateway to WebRTC started.  More needed.

12.20.

11.21.  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.

12.21.

11.22.  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 multiparty-unaware endpoints

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

12.22.

11.23.  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 multiparty-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 multiparty-aware endpoints, and a
   new section on the multiparty-unaware mixing with low functionality
   but SHOULD 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 whose text is included in one or more text
   blocks.  It is not the same as the list of participants in a
   conference.  With audio and video media, the CSRC-list would often
   contain all participants who are not muted whereas text participants
   that don't type are completely silent and so don't show up in RTP
   packet CSRC-lists.

12.23.

11.24.  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.

12.  References

13.1.

12.1.  Normative References

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

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630,
              DOI 10.17487/RFC5630, October 2009,
              <https://www.rfc-editor.org/info/rfc5630>.

   [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>.

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

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

   [RFC8865]  Holmberg, C. and G. Hellström, "T.140 Real-Time Text
              Conversation over WebRTC Data Channels", RFC 8865,
              DOI 10.17487/RFC8865, January 2021,
              <https://www.rfc-editor.org/info/rfc8865>.

   [RFC8866]  Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP:
              Session Description Protocol", RFC 8866,
              DOI 10.17487/RFC8866, January 2021,
              <https://www.rfc-editor.org/info/rfc8866>.

   [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.

12.2.  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>.

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

   [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>.

   [RFC8723]  Jennings, C., Jones, P., Barnes, R., and A.B. Roach,
              "Double Encryption Procedures for the Secure Real-Time
              Transport Protocol (SRTP)", RFC 8723,
              DOI 10.17487/RFC8723, April 2020,
              <https://www.rfc-editor.org/info/rfc8723>.

   [RFC8825]  Alvestrand, H., "Overview: Real-Time Protocols for
              Browser-Based Applications", RFC 8825,
              DOI 10.17487/RFC8825, January 2021,
              <https://www.rfc-editor.org/info/rfc8825>.

Acknowledgements

   The author want to thank the following persons for support, reviews
   and valuable comments: Bernard Aboba, Amanda Baber, Roman Danyliw,
   Spencer Dawkins, Martin Duke, Lars Eggert, James Hamlin, Benjamin
   Kaduk, Murray Kucherawy, Paul Kyziwat, Jonathan Lennox, Lorenzo
   Miniero, Dan Mongrain, Francesca Palombini, Colin Perkins, Brian
   Rosen, Juergen Schoenwaelder, Rich Salz, Robert Wilton, Dale Worley,
   Peter Yee and Yong Xin.

Author's Address
   Gunnar Hellstrom
   Gunnar Hellstrom Accessible Communication
   SE-13670 Vendelso
   Sweden

   Email: gunnar.hellstrom@ghaccess.se