draft-ietf-avtcore-multi-party-rtt-mix-11.txt   draft-ietf-avtcore-multi-party-rtt-mix-12.txt 
AVTCore G. Hellstrom AVTCore G. Hellstrom
Internet-Draft Gunnar Hellstrom Accessible Communication Internet-Draft Gunnar Hellstrom Accessible Communication
Updates: RFC 4103 (if approved) 22 November 2020 Updates: 4103 (if approved) 15 December 2020
Intended status: Standards Track Intended status: Standards Track
Expires: 26 May 2021 Expires: 18 June 2021
RTP-mixer formatting of multi-party Real-time text RTP-mixer formatting of multi-party Real-time text
draft-ietf-avtcore-multi-party-rtt-mix-11 draft-ietf-avtcore-multi-party-rtt-mix-12
Abstract Abstract
Real-time text mixers for multi-party sessions need to identify the Real-time text mixers for multi-party sessions need to identify the
source of each transmitted group of text so that the text can be source of each transmitted group of text so that the text can be
presented by endpoints in suitable grouping with other text from the presented by endpoints in suitable grouping with other text from the
same source, while new text from other sources is also presented in same source, while new text from other sources is also presented in
readable grouping as received interleaved in real-time. readable grouping as received interleaved in real-time.
Regional regulatory requirements specify provision of real-time text Use of RTT is increasing, and specifically, use in emergency calls is
in multi-party calls. RFC 4103 mixer implementations can use increasing. Emergency call use requires multi-party mixing. RFC
traditional RTP functions for source identification, but the mixer 4103 "RTP Payload for Text Conversation" mixer implementations can
source switching performance is limited when using the default use traditional RTP functions for source identification, but the
transmission characteristics with redundancy. performance of the mixer when giving turns for the different sources
to transmit is limited when using the default transmission
characteristics with redundancy.
Enhancements for RFC 4103 real-time text mixing is provided in this Enhancements for RFC 4103 real-time text mixing are provided in this
document, suitable for a centralized conference model that enables document, suitable for a centralized conference model that enables
source identification and source switching. The intended use is for source identification and rapidly interleaved transmission of text
real-time text mixers and multi-party-aware participant endpoints. from different sources. The intended use is for real-time text
The specified mechanism build on the standard use of the CSRC list in mixers and participant endpoints capable of providing an efficient
the RTP packet for source identification. The method makes use of presentation or other treatment of a multi-party real-time text
the same "text/t140" and "text/red" formats as for two-party session. The specified mechanism builds on the standard use of the
sessions. CSRC list in the RTP packet for source identification. The method
makes use of the same "text/t140" and "text/red" formats as for two-
party sessions.
Solutions using multiple RTP streams in the same RTP session are
briefly mentioned, as they could have some benefits over the RTP-
mixer model. The possibility to implement the solution in a wide
range of existing RTP implementations made the RTP-mixer model be
selected to be fully specified in this document.
A capability exchange is specified so that it can be verified that a A capability exchange is specified so that it can be verified that a
participant can handle the multi-party coded real-time text stream. mixer and a participant can handle the multi-party coded real-time
The capability is indicated by use of a media attribute "rtt-mixer". text stream using the RTP-mixer method. The capability is indicated
by use of an SDP media attribute "rtt-mixer".
The document updates RFC 4103[RFC4103] The document updates RFC 4103 "RTP Payload for Text Conversation".
A specifications of how a mixer can format text for the case when the A specification of how a mixer can format text for the case when the
endpoint is not multi-party aware is also provided. endpoint is not multi-party aware is also provided.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on 26 May 2021. This Internet-Draft will expire on 18 June 2021.
Copyright Notice Copyright Notice
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Selected solution and considered alternative . . . . . . 5 1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6
1.2. Nomenclature . . . . . . . . . . . . . . . . . . . . . . 7 1.2. Selected solution and considered alternatives . . . . . . 6
1.3. Intended application . . . . . . . . . . . . . . . . . . 8 1.3. Intended application . . . . . . . . . . . . . . . . . . 9
2. Overview over the two specified solutions . . . . . . . . . . 9 2. Overview of the two specified solutions and selection of
2.1. Negotiated use of the RFC 4103 format for multi-party method . . . . . . . . . . . . . . . . . . . . . . . . . 10
transmission in a single RTP stream . . . . . . . . . . . 9 2.1. The RTP-mixer based solution for multi-party aware
2.2. Mixing for multi-party unaware endpoints . . . . . . . . 9 endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10
3. Details for the multi-party aware mixing case . . . . . . . . 9 2.2. Mixing for multi-party unaware endpoints . . . . . . . . 10
3.1. Offer/answer considerations . . . . . . . . . . . . . . . 10 2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11
3.2. Actions depending on capability negotiation result . . . 10 2.4. Actions depending on capability negotiation result . . . 11
3.3. Use of fields in the RTP packets . . . . . . . . . . . . 10 3. Details for the RTP-mixer based multi-party aware mixing
3.4. Initial transmission of a BOM character . . . . . . . . . 11 method . . . . . . . . . . . . . . . . . . . . . . . . . 12
3.5. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 11 3.1. Use of fields in the RTP packets . . . . . . . . . . . . 12
3.6. Transmission interval . . . . . . . . . . . . . . . . . . 11 3.2. Initial transmission of a BOM character . . . . . . . . . 12
3.7. Only one source per packet . . . . . . . . . . . . . . . 11 3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 12
3.8. Do not send received text to the originating source . . . 12 3.4. Transmission interval . . . . . . . . . . . . . . . . . . 13
3.9. Clean incoming text . . . . . . . . . . . . . . . . . . . 12 3.5. Only one source per packet . . . . . . . . . . . . . . . 13
3.10. Redundant transmission principles . . . . . . . . . . . . 12 3.6. Do not send received text to the originating source . . . 13
3.11. Interleaving text from different sources . . . . . . . . 12 3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 13
3.12. Text placement in packets . . . . . . . . . . . . . . . . 12 3.8. Redundant transmission principles . . . . . . . . . . . . 13
3.13. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 13 3.9. Interleaving text from different sources . . . . . . . . 14
3.14. Creation of the redundancy . . . . . . . . . . . . . . . 13 3.10. Text placement in packets . . . . . . . . . . . . . . . . 14
3.15. Timer offset fields . . . . . . . . . . . . . . . . . . . 14 3.11. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 15
3.16. Other RTP header fields . . . . . . . . . . . . . . . . . 14 3.12. Creation of the redundancy . . . . . . . . . . . . . . . 15
3.17. Pause in transmission . . . . . . . . . . . . . . . . . . 14 3.13. Timer offset fields . . . . . . . . . . . . . . . . . . . 15
3.18. RTCP considerations . . . . . . . . . . . . . . . . . . . 15 3.14. Other RTP header fields . . . . . . . . . . . . . . . . . 16
3.19. Reception of multi-party contents . . . . . . . . . . . . 15 3.15. Pause in transmission . . . . . . . . . . . . . . . . . . 16
3.20. Performance considerations . . . . . . . . . . . . . . . 17 3.16. RTCP considerations . . . . . . . . . . . . . . . . . . . 16
3.21. Security for session control and media . . . . . . . . . 18 3.17. Reception of multi-party contents . . . . . . . . . . . . 16
3.22. SDP offer/answer examples . . . . . . . . . . . . . . . . 18 3.18. Performance considerations . . . . . . . . . . . . . . . 18
3.23. Packet sequence example from interleaved transmission . . 19 3.19. Security for session control and media . . . . . . . . . 19
3.24. Maximum character rate "CPS" . . . . . . . . . . . . . . 22 3.20. SDP offer/answer examples . . . . . . . . . . . . . . . . 19
4. Presentation level considerations . . . . . . . . . . . . . . 22 3.21. Packet sequence example from interleaved transmission . . 20
4.1. Presentation by multi-party aware endpoints . . . . . . . 23 3.22. Maximum character rate "CPS" . . . . . . . . . . . . . . 23
4.2. Multi-party mixing for multi-party unaware endpoints . . 25 4. Presentation level considerations . . . . . . . . . . . . . . 23
5. Relation to Conference Control . . . . . . . . . . . . . . . 30 4.1. Presentation by multi-party aware endpoints . . . . . . . 24
5.1. Use with SIP centralized conferencing framework . . . . . 31 4.2. Multi-party mixing for multi-party unaware endpoints . . 26
5.2. Conference control . . . . . . . . . . . . . . . . . . . 31 5. Relation to Conference Control . . . . . . . . . . . . . . . 31
6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 31 5.1. Use with SIP centralized conferencing framework . . . . . 32
6.1. Gateway considerations with Textphones (e.g. TTYs). . . 31 5.2. Conference control . . . . . . . . . . . . . . . . . . . 32
6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 32 6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 32
7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 32 6.1. Gateway considerations with Textphones (e.g. TTYs). . . 32
8. Congestion considerations . . . . . . . . . . . . . . . . . . 32 6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 33
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33 7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 34
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33 8. Congestion considerations . . . . . . . . . . . . . . . . . . 34
10.1. Registration of the "rtt-mixer" sdp media attribute . . 33 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 34
11. Security Considerations . . . . . . . . . . . . . . . . . . . 34 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 34
12. Change history . . . . . . . . . . . . . . . . . . . . . . . 34 10.1. Registration of the "rtt-mixer" sdp media attribute . . 34
11. Security Considerations . . . . . . . . . . . . . . . . . . . 35
12. Change history . . . . . . . . . . . . . . . . . . . . . . . 35
12.1. Changes included in 12.1. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 34 draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 35
12.2. Changes included in 12.2. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 34 draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 36
12.3. Changes included in 12.3. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 34 draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 36
12.4. Changes included in 12.4. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 35 draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 36
12.5. Changes included in 12.5. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 35 draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 36
12.6. Changes included in 12.6. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 35 draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 37
12.7. Changes included in 12.7. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 35 draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 37
12.8. Changes included in 12.8. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 36 draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 37
12.9. Changes included in 12.9. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 36 draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 37
12.10. Changes included in 12.10. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 37 draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 37
12.11. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 37 12.11. Changes included in
12.12. Changes from draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 38
draft-hellstrom-avtcore-multi-party-rtt-source-03 to 12.12. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 39
draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 37
12.13. Changes from 12.13. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-02 to draft-hellstrom-avtcore-multi-party-rtt-source-03 to
-03 . . . . . . . . . . . . . . . . . . . . . . . . . . 38 draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 39
12.14. Changes from 12.14. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-01 to draft-hellstrom-avtcore-multi-party-rtt-source-02 to
-02 . . . . . . . . . . . . . . . . . . . . . . . . . . 38 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . 39
12.15. Changes from 12.15. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-01 to
-02 . . . . . . . . . . . . . . . . . . . . . . . . . . 40
12.16. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-00 to draft-hellstrom-avtcore-multi-party-rtt-source-00 to
-01 . . . . . . . . . . . . . . . . . . . . . . . . . . 39 -01 . . . . . . . . . . . . . . . . . . . . . . . . . . 40
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 41
13.1. Normative References . . . . . . . . . . . . . . . . . . 39 13.1. Normative References . . . . . . . . . . . . . . . . . . 41
13.2. Informative References . . . . . . . . . . . . . . . . . 40 13.2. Informative References . . . . . . . . . . . . . . . . . 42
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 41 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 43
1. Introduction 1. Introduction
RFC 4103[RFC4103] specifies use of RFC 3550 RTP [RFC3550] for "RTP Payload for Text Conversation" [RFC4103] specifies use of RFC
transmission of real-time text (RTT) and the "text/t140" format. It 3550 RTP [RFC3550] for transmission of real-time text (RTT) and the
also specifies a redundancy format "text/red" for increased "text/t140" format. It also specifies a redundancy format "text/red"
robustness. RFC 4102 [RFC4102] registers the "text/red" format. for increased robustness. RFC 4102 [RFC4102] registers the "text/
Regional regulatory requirements specify provision of real-time text red" format.
in multi-party calls.
Real-time text is usually provided together with audio and sometimes Real-time text is usually provided together with audio and sometimes
with video in conversational sessions. with video in conversational sessions.
A requirement related to multi-party sessions from the presentation A requirement related to multi-party sessions from the presentation
level standard T.140 for real-time text is: "The display of text from level standard T.140 for real-time text is: "The display of text from
the members of the conversation should be arranged so that the text the members of the conversation should be arranged so that the text
from each participant is clearly readable, and its source and the from each participant is clearly readable, and its source and the
relative timing of entered text is visualized in the display." relative timing of entered text is visualized in the display."
Another requirement is that the mixing procedure must not introduce Another requirement is that the mixing procedure must not introduce
delays in the text streams that are experienced disturbing the real- delays in the text streams that are experienced to be disturbing the
time experience of the receiving users. real-time experience of the receiving users.
The redundancy scheme of RFC 4103 [RFC4103] enables efficient The redundancy scheme of RFC 4103 [RFC4103] enables efficient
transmission of redundant text in packets together with new text. transmission of earlier transmitted redundant text in packets
However the redundancy header format has no source indicators for the together with new text. However the redundancy header format has no
redundant transmissions. The redundant parts in a packet must source indicators for the redundant transmissions. The redundant
therefore be from the same source as the new text. The recommended parts in a packet must therefore be from the same source as the new
transmission is one new and two redundant generations of text text. The recommended transmission is one new and two redundant
(T140blocks) in each packet and the recommended transmission interval generations of text (T140blocks) in each packet and the recommended
for two-party use is 300 ms. transmission interval for two-party use is 300 ms.
Real-time text mixers for multi-party sessions therefore need to Real-time text mixers for multi-party sessions need to include the
insert the source of each transmitted group of text from a conference source with each transmitted group of text from a conference
participant so that the text can be transmitted interleaved with text participant so that the text can be transmitted interleaved with text
groups from different sources in the rate they are created. This groups from different sources in the rate they are created. This
enables the text groups to be presented by endpoints in suitable enables the text groups to be presented by endpoints in suitable
grouping with other text from the same source. The presentation can grouping with other text from the same source.
then be arranged so that text from different sources can be presented
in real-time and easily read while it is possible for a reading user
to also perceive approximately when the text was created in real time
by the different parties. The transmission and mixing is intended to
be done in a general way so that presentation can be arranged in a
layout decided by the endpoint.
There are existing implementations of RFC 4103 without the updates The presentation can then be arranged so that text from different
from this document. These will not be able to receive and present sources can be presented in real-time and easily read. At the same
real-time text mixed for multi-party aware endpoints. 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 multi-party aware endpoints.
A negotiation mechanism is therefore needed for verification if the A negotiation mechanism is therefore needed for verification if the
parties are able to handle a multi-party coded stream and agreeing on parties are able to handle a common method for multi-party
using that method. transmission and agreeing on using that method.
A fall-back mixing procedure is also needed for cases when the A fall-back mixing procedure is also needed for cases when the
negotiation result indicates that a receiving endpoint is not capable negotiation result indicates that a receiving endpoint is not capable
of handling the mixed format. This method is called the mixing of handling the mixed format. Multi-party unaware endpoints would
procedure for multi-party unaware endpoints. The fall-back method is possibly otherwise present all received multi-party mixed text as if
naturally not expected to meet all performance requirements placed on it came from the same source regardless of any accompanying source
the mixing procedure for multi-party aware endpoints. indication coded in fields in the packet. Or they may have any other
undesirable way of acting on the multi-party content. The fall-back
method is called the mixing procedure for multi-party unaware
endpoints. The fall-back method is naturally not expected to meet
all performance requirements placed on the mixing procedure for
multi-party aware endpoints.
The document updates RFC 4103[RFC4103] by introducing an attribute The document updates RFC 4103[RFC4103] by introducing an attribute
for indicating capability for the multi-party mixing case and rules for indicating capability for the RTP-mixer based multi-party mixing
for source indications and source switching. case and rules for source indications and interleaving of text from
different sources.
1.1. Selected solution and considered alternative 1.1. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP-
mixer, RTP-translator as defined in [RFC3550]
The term "T140block" is defined in RFC 4103 [RFC4103] to contain one
or more T.140 code elements.
"TTY" stands for a text telephone type used in North America.
"WebRTC" stands for web based communication specified by W3C and
IETF.
"DTLS-SRTP" stands for security specified in RFC 5764 [RFC5764].
"multi-party aware" stands for an endpoint receiving real-time text
from multiple sources through a common conference mixer being able to
present the text in real-time separated by source and presented so
that a user can get an impression of the approximate relative timing
of text from different parties.
"multi-party unaware" stands for an endpoint not itself being able to
separate text from different sources when received through a common
conference mixer.
1.2. Selected solution and considered alternatives
A number of alternatives were considered when searching an efficient A number of alternatives were considered when searching an efficient
and easily implemented multi-party method for real-time text. This and easily implemented multi-party method for real-time text. This
section explains a few of them briefly. section explains a few of them briefly.
One RTP stream per source, sent in the same RTP session with Multiple RTP streams, one per participant.
"text/red" format. One RTP stream per source would be sent in the same RTP session
From some points of view, use of multiple RTP streams, one for with the "text/red" format. From some points of view, use of
each source, sent in the same RTP session, called the RTP multiple RTP streams, one for each source, sent in the same RTP
translator model in [RFC3550], would be efficient, and use exactly session would be efficient, and would use exactly the same packet
the same packet format as [RFC4103], the same payload type and a format as [RFC4103] and the same payload type. A couple of
simple SDP declaration. However, the RTP implementation in both relevant scenarios using multiple RTP-streams are specified in
mixers and endpoints need to support multiple streams in the same "RTP Topologies" [RFC7667]. One possibility of special interest
RTP session in order to use this mechanism. For best deployment is the Selective Forwarding Middlebox (SFM) topology specified in
opportunity, it should be possible to upgrade existing endpoint RFC 7667 section 3.7 that could enable end to end encryption. In
solutions to be multi-party aware with a reasonable effort. There contrast to audio and video, real-time text is only transmitted
is currently a lack of support for multi-stream RTP in certain when the users actually transmit information. Thus an SFM
implementation technologies. This fact made this solution not solution would not need to exclude any party from transmission
selected for inclusion in this document. under normal conditions. In order to allow the mixer to convey
the packets with the payload preserved and encrypted, an SFM
solution would need to act on some specific characteristics of the
"text/red" format. The redundancy headers are part of the
payload, so the receiver would need to just assume that the
payload type number in the redundancy header is for "text/t140".
The characters per second parameter (CPS) would need to act per
stream. The relation between the SSRC and the source would need
to be conveyed in some specified way, e.g. in the CSRC. Recovery
and loss detection would preferably be based on sequence number
gap detection. Thus sequence number gaps in the incoming stream
to the mixer would need to be reflected in the stream to the
participant and no new gaps created by the mixer. However, the
RTP implementation in both mixers and endpoints need to support
multiple streams in the same RTP session in order to use this
mechanism. For best deployment opportunity, it should be possible
to upgrade existing endpoint solutions to be multi-party aware
with a reasonable effort. There is currently a lack of support
for multi-stream RTP in certain implementation technologies. This
fact made this solution only briefly mentioned in this document as
an option for further study.
The "text/red" format in RFC 4103 with shorter transmission RTP-mixer based method for multi-party aware endpoints.
interval, and indicating source in CSRC. The "text/red" format in RFC 4103 is sent with shorter
The "text/red" format with "text/t140" payload in a single RTP transmission interval with the RTP-mixer method and indicating
stream can be sent with 100 ms packet intervals instead of the source in CSRC. The "text/red" format with "text/t140" payload in
regular 300 ms. The source is indicated in the CSRC field. a single RTP stream can be sent with 100 ms packet intervals
Source switching can then be done every 100 ms while simultaneous instead of the regular 300 ms. The source is indicated in the
CSRC field. Transmission of packets with text from different
sources can then be done every 100 ms while simultaneous
transmission occurs. With five participants sending text transmission occurs. With five participants sending text
simultaneously, the switching and transmission performance is simultaneously, the switching and transmission performance is
good. With more simultaneously sending participants, there will good. With more simultaneously sending participants, there will
be a noticable jerkiness in text presentation. The jerkiness will be a noticeable jerkiness in text presentation. The jerkiness
be more expressed the more participants who send text will be more expressed the more participants who send text
simultaneously. With ten sending participants, the jerkiness will simultaneously. With ten sending participants, the jerkiness will
be about one second. Text sent from a source at the end of the be about one second. Text sent from a source at the end of the
period its text is sent by the mixer will have close to zero extra period its text is sent by the mixer will have close to zero extra
delay. Recent text will be presented with no or low delay. The delay. Recent text will be presented with no or low delay. The
one second jerkiness will be noticable and slightly unpleasant, one second jerkiness will be noticeable and slightly unpleasant,
but corresponds in time to what typing humans often cause by but corresponds in time to what typing humans often cause by
hesitation or changing position while typing. A benefit of this hesitation or changing position while typing. A benefit of this
method is that no new packet format needs to be introduced and method is that no new packet format needs to be introduced and
implemented. Since simultaneous typing by more than two parties implemented. Since simultaneous typing by more than two parties
is very rare, and in most applications also more than three is very rare, this method can be used successfully with good
parties in a call is rare, this method can be used successfully performance. Recovery of text in case of packet loss is based on
with good performance. Recovery of text in case of packet loss is analysis of timestamps of received redundancy versus earlier
based on analysis of timestamps of received redundancy versus received text. Negotiation is based on a new sdp media attribute
earlier received text. Negotiation is based on a new sdp media "rtt-mixer". This method is selected to be the main one specified
attribute "rtt-mixer". This method is selected to be the main one in this document.
specified in this document.
A new "text" media subtype with up to 15 sources in each packet. Multiple sources per packet.
The mechanism makes use of the RTP mixer model specified in A new "text" media subtype would be specified with up to 15
RFC3550[RFC3550]. Text from up to 15 sources can be included in sources in each packet. The mechanism would make use of the RTP
each packet. Packets are normally sent every 300 ms. The mean mixer model specified in RFC3550[RFC3550]. Text from up to 15
delay will be 150 ms. The sources are indicated in strict order sources can be included in each packet. Packets are normally sent
in the CSRC list of the RTP packets. A new redundancy packet every 300 ms. The mean delay will be 150 ms. The sources are
format is specified. This method would result in good indicated in strict order in the CSRC list of the RTP packets. A
performance, but would require standardisation and implementation new redundancy packet format is specified. This method would
of new releases in the target technologies that would take more result in good performance, but would require standardisation and
time than desirable to complete. It was therefore not selected to implementation of new releases in the target technologies that
be included in this document. would take more time than desirable to complete. It was therefore
not selected to be included in this document.
The presentation planned by the mixer for multi-party unaware Mixing for multi-party unaware endpoints
endpoints. Presentation of text from multiple parties is prepared by the
It is desirable to have a method that does not require any mixer in one single stream. It is desirable to have a method that
modifications in existing user devices implementing RFC 4103 for does not require any modifications in existing user devices
RTT without explicit support of multi-party sessions. This is implementing RFC 4103 for RTT without explicit support of multi-
possible by having the mixer insert a new line and a text party sessions. This is possible by having the mixer insert a new
formatted source label before each switch of text source in the line and a text formatted source label before each switch of text
stream. Switch of source can only be done in places in the text source in the stream. Switch of source can only be done in places
where it does not disturb the perception of the contents. Text in the text where it does not disturb the perception of the
from only one source can be presented in real time at a time. The contents. Text from only one source can be presented in real time
delay will therefore be varying. The method has also other at a time. The delay will therefore be varying. The method also
limitations, but is included in this document as a fallback has other limitations, but is included in this document as a
method. In calls where parties take turns properly by ending fallback method. In calls where parties take turns properly by
their entries with a new line, the limitations will have limited ending their entries with a new line, the limitations will have
influence on the user experience. while only two parties send limited influence on the user experience. while only two parties
text, these two will see the text in real time with no delay. 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. This method is specified as a fallback method in this document.
RTT transport in WebRTC RTT transport in WebRTC
Transport of real-time text in the WebRTC technology is specified Transport of real-time text in the WebRTC technology is specified
to use the WebRTC data channel in to use the WebRTC data channel in
[I-D.ietf-mmusic-t140-usage-data-channel]. That spcification [I-D.ietf-mmusic-t140-usage-data-channel]. That specification
contains a section briefly describing its use in multi-party contains a section briefly describing its use in multi-party
sessions. The focus of this document is RTP transport. sessions. The focus of this document is RTP transport.
Therefore, even if the WebRTC transport provides good multi-party Therefore, even if the WebRTC transport provides good multi-party
performance, it is just mentioned in this document in relation to performance, it is just mentioned in this document in relation to
providing gateways with multi-party capabilities between RTP and providing gateways with multi-party capabilities between RTP and
WebRTC technologies. WebRTC technologies.
1.2. Nomenclature
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP-
mixer, RTP-translator are explained in [RFC3550]
The term "T140block" is defined in RFC 4103 [RFC4103] to contain one
or more T.140 code elements.
"TTY" stands for a text telephone type used in North America.
"WebRTC" stands for web based communication specified by W3C and
IETF.
"DTLS-SRTP" stands for security specified in RFC 5764 [RFC5764].
"multi-party aware" stands for an endpoint receiving real-time text
from multiple sources through a common conference mixer being able to
present the text in real-time separated by source and presented so
that a user can get an impression of the approximate relative timing
of text from different parties.
"multi-party unaware" stands for an endpoint not itself being able to
separate text from different sources when received through a common
conference mixer.
1.3. Intended application 1.3. Intended application
The method for multi-party real-time text specified in this document The method for multi-party real-time text specified in this document
is primarily intended for use in transmission between mixers and is primarily intended for use in transmission between mixers and
endpoints in centralised mixing configurations. It is also endpoints in centralised mixing configurations. It is also
applicable between mixers. An often mentioned application is for applicable between mixers. An often mentioned application is for
emergency service calls with real-time text and voice, where a emergency service calls with real-time text and voice, where a
calltaker want to make an attended handover of a call to another calltaker want to make an attended handover of a call to another
agent, and stay observing the session. Multimedia conference agent, and stay observing the session. Multimedia conference
sessions with support for participants to contribute in text is sessions with support for participants to contribute in text is
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text conversion is yet another mentioned application. text conversion is yet another mentioned application.
In all these applications, normally only one participant at a time In all these applications, normally only one participant at a time
will send long text utterances. In some cases, one other participant will send long text utterances. In some cases, one other participant
will occasionally contribute with a longer comment simultaneously. will occasionally contribute with a longer comment simultaneously.
That may also happen in some rare cases when text is interpreted to That may also happen in some rare cases when text is interpreted to
text in another language in a conference. Apart from these cases, text in another language in a conference. Apart from these cases,
other participants are only expected to contribute with very brief other participants are only expected to contribute with very brief
utterings while others are sending text. 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, Text is supposed to be human generated, by some text input means,
such as typing on a keyboard or using speech-to-text technology. such as typing on a keyboard or using speech-to-text technology.
Occasional small cut-and-paste operations may appear even if that is Occasional small cut-and-paste operations may appear even if that is
not the initial purpose of real-time text. not the initial purpose of real-time text.
The real-time characteristics of real-time text is essential for the The real-time characteristics of real-time text is essential for the
participants to be able to contribute to a conversation. If the text participants to be able to contribute to a conversation. If the text
is too much delayed from typing a letter to its presentation, then, is too much delayed from typing a letter to its presentation, then,
in some conference situations, the opportunity to comment will be in some conference situations, the opportunity to comment will be
gone and someone else will grab the turn. A delay of more than one gone and someone else will grab the turn. A delay of more than one
second in such situations is an obstacle for good conversation. second in such situations is an obstacle for good conversation.
2. Overview over the two specified solutions 2. Overview of the two specified solutions and selection of method
This section contains a brief introduction of the two methods This section contains a brief introduction of the two methods
specified in this document. specified in this document.
2.1. Negotiated use of the RFC 4103 format for multi-party transmission 2.1. The RTP-mixer based solution for multi-party aware endpoints
in a single RTP stream
The main purpose of this document is to specify a method for true This method specifies negotiated use of the RFC 4103 format for
multi-party real-time text mixing for multi-party aware endpoints. multi-party transmission in a single RTP stream. The main purpose of
The method makes use of the current format for real-time text in this document is to specify a method for true multi-party real-time
[RFC4103]. It is an update of RFC 4103 by a clarification on one way text mixing for multi-party aware endpoints that can be widely
to use it in the multi-party situation. It is done by completing a deployed. The RTP-mixer based method makes use of the current format
negotiation for this kind of multi-party capability and by indicating for real-time text in [RFC4103]. It is an update of RFC 4103 by a
source in the CSRC element in the RTP packets. Specific clarification on one way to use it in the multi-party situation.
That is done by completing a negotiation for this kind of multi-party
capability and by interleaving packets from different sources. The
source is indicated in the CSRC element in the RTP packets. Specific
considerations are made to be able to recover text after packet loss. considerations are made to be able to recover text after packet loss.
The detailed procedures for the multi-party aware case are specified The detailed procedures for the RTP-mixer based multi-party aware
in Section 3 case are specified in Section 3.
Please use [RFC4103] as reference when reading the specification. Please use [RFC4103] as reference when reading the specification.
2.2. Mixing for multi-party unaware endpoints 2.2. Mixing for multi-party unaware endpoints
A method is also specified in this document for cases when the A method is also specified in this document for cases when the
endpoint participating in a multi-party call does not itself endpoint participating in a multi-party call does not itself
implement any solution for multi-party handling of real-time text. implement any solution, or not the same, as the mixer. The method
The solution requires the mixer to insert text dividers and readable requires the mixer to insert text dividers and readable labels and
labels and only send text from one source at a time until a suitable only send text from one source at a time until a suitable point
point appears for source change. This solution is a fallback method appears for source change. This solution is a fallback method with
with functional limitations. It acts on the presentation level. functional limitations. It acts on the presentation level.
A party performing as a mixer, which has not negotiated the "rtt- A party acting as a mixer, which has not negotiated any method for
mixer" sdp media attribute, but negotiated a "text/red" or "text/ true multi-party RTT handling, but negotiated a "text/red" or "text/
t140" format in a session with a participant SHOULD, if nothing else t140" format in a session with a participant SHOULD, if nothing else
is specified for the application, format transmitted text to that is specified for the application, format transmitted text to that
participant to be suitable to present on a multi-party unaware participant to be suitable to present on a multi-party unaware
endpoint as further specified in Section 4.2. endpoint as further specified in Section 4.2.
3. Details for the multi-party aware mixing case 2.3. Offer/answer considerations
3.1. Offer/answer considerations
RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a
redundancy format "text/red" for increased robustness of real-time redundancy format "text/red" for increased robustness of real-time
text transmission. This document updates RFC 4103[RFC4103] by text transmission. This document updates RFC 4103[RFC4103] by
introducing a capability negotiation for handling multi-party real- introducing a capability negotiation for handling multi-party real-
time text, a way to indicate the source of transmitted text, and time text, a way to indicate the source of transmitted text, and
rules for efficient timing of the transmissions interleaved from rules for efficient timing of the transmissions interleaved from
different sources. different sources.
The capability negotiation is based on use of the sdp media attribute The capability negotiation for the "RTP-mixer based multi-party
"rtt-mixer". method" is based on use of the sdp media attribute "rtt-mixer".
Both parties shall indicate their capability in a session setup or Both parties SHALL indicate their capability in a session setup or
modification, and evaluate the capability of the counterpart. modification, and evaluate the capability of the counterpart.
The syntax is as follows: The syntax is as follows:
"a=rtt-mixer" "a=rtt-mixer"
3.2. Actions depending on capability negotiation result If any other method for RTP-based multi-party real-time text gets
specified, it is assumed that it will be recognized by some specific
SDP feature exchange.
A transmitting party SHALL send text according to the multi-party It is possible to both indicate capability for the RTP-mixer based
format only when the negotiation for this method was successful and method and another method. An answer MUST NOT accept more than one
when the CC field in the RTP packet is set to 1. In all other cases, method.
the packets SHALL be populated and interpreted as for a two-party
session. 2.4. Actions depending on capability negotiation result
A transmitting party SHALL send text according to the RTP-mixer based
multi-party method only when the negotiation for that method was
successful and when it conveys text for another source. In all other
cases, the packets SHALL be populated and interpreted as for a two-
party session.
A party which has negotiated the "rtt-mixer" sdp media attribute MUST A party which has negotiated the "rtt-mixer" sdp media attribute MUST
populate the CSRC-list and format the packets according to Section 3 populate the CSRC-list and format the packets according to Section 3
if it acts as an rtp-mixer and sends multi-party text. if it acts as an rtp-mixer and sends multi-party text.
A party which has negotiated the "rtt-mixer" sdp media attribute MUST A party which has negotiated the "rtt-mixer" sdp media attribute MUST
interpret the contents of the "CC" field the CSRC-list and the interpret the contents of the "CC" field, the CSRC-list and the
packets according to Section 3 in received rtp packets in the packets according to Section 3 in received RTP packets in the
corresponding RTP stream. corresponding RTP stream.
A party not performing as a mixer MUST not include the CSRC list. A party which has not successfully completed the negotiation of the
"rtt-mixer" sdp media attribute MUST NOT transmit packets interleaved
from different sources in the same RTP stream as specified in
Section 3. If the party is a mixer and did declare the "rtt-mixer"
sdp media attribute, it SHOULD perform the procedure for multi-party
unaware endpoints. If the party is not a mixer, it SHOULD transmit
according to [RFC4103].
3.3. Use of fields in the RTP packets 3. Details for the RTP-mixer based multi-party aware mixing method
3.1. Use of fields in the RTP packets
The CC field SHALL show the number of members in the CSRC list, which The CC field SHALL show the number of members in the CSRC list, which
SHALL be one (1) in transmissions from a mixer involved in a multi- SHALL be one (1) in transmissions from a mixer when conveying text
party session, and otherwise 0. from other sources in a multi-party session, and otherwise 0.
When transmitted from a mixer during a multi-party session, a CSRC When text is conveyed by a mixer during a multi-party session, a CSRC
list SHALL be included in the packet. The single member in the CSRC- list SHALL be included in the packet. The single member in the CSRC-
list SHALL contain the SSRC of the source of the T140blocks in the list SHALL contain the SSRC of the source of the T140blocks in the
packet. When redundancy is used, the recommended level of redundancy packet.
is to use one primary and two redundant generations of T140blocks.
In some cases, a primary or redundant T140block is empty, but is When redundancy is used, the RECOMMENDED level of redundancy is to
still represented by a member in the redundancy header. use one primary and two redundant generations of T140blocks. In some
cases, a primary or redundant T140block is empty, but is still
represented by a member in the redundancy header.
From other aspects, the contents of the RTP packets are equal to what From other aspects, the contents of the RTP packets are equal to what
is specified in [RFC4103]. is specified in [RFC4103].
3.4. Initial transmission of a BOM character 3.2. Initial transmission of a BOM character
As soon as a participant is known to participate in a session with As soon as a participant is known to participate in a session with
another entity and being available for text reception, a Unicode BOM another entity and being available for text reception, a Unicode BOM
character SHALL be sent to it by the other entity according to the character SHALL be sent to it by the other entity according to the
procedures in this section. If the transmitter is a mixer, then the procedures in this section. If the transmitter is a mixer, then the
source of this character SHALL be indicated to be the mixer itself. source of this character SHALL be indicated to be the mixer itself.
Note that the BOM character SHALL be transmitted with the same Note that the BOM character SHALL be transmitted with the same
redundancy procedures as any other text. redundancy procedures as any other text.
3.5. Keep-alive 3.3. Keep-alive
After that, the transmitter SHALL send keep-alive traffic to the After that, the transmitter SHALL send keep-alive traffic to the
receiver(s) at regular intervals when no other traffic has occurred receiver(s) at regular intervals when no other traffic has occurred
during that interval, if that is decided for the actual connection. during that interval, if that is decided for the actual connection.
Recommendations for keep-alive can be found in [RFC6263]. 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.
3.6. Transmission interval 3.4. Transmission interval
A "text/red" or "text/t140" transmitter in a mixer SHOULD send A "text/red" or "text/t140" transmitter in a mixer SHOULD send
packets distributed in time as long as there is something (new or packets distributed in time as long as there is something (new or
redundant T140blocks) to transmit. The maximum transmission interval redundant T140blocks) to transmit. The maximum transmission interval
SHOULD then be 330 ms. It is RECOMMENDED to send next packet to a SHOULD then be 330 ms. It is RECOMMENDED to send the next packet to
receiver as soon as new text to that receiver is available, as long 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 as the time after the latest sent packet to the same receiver is more
than or equal to 100 ms, and also the maximum character rate ("CPS") than or equal to 100 ms, and also the maximum character rate ("CPS")
to the receiver is not exceeded. The intention is to keep the to the receiver is not exceeded. The intention is to keep the
latency low and network load limited while keeping a good protection latency low and network load limited while keeping a good protection
against text loss in bursty packet loss conditions. against text loss in bursty packet loss conditions.
For a transmitter not acting in a mixer, the same transmission For a transmitter not acting in a mixer, the same transmission
interval principles apply, but the maximum transmission interval interval principles apply, but the maximum transmission interval
SHOULD be 300 ms. SHOULD be 300 ms.
3.7. Only one source per packet 3.5. Only one source per packet
New and redundant text from one source SHALL be transmitted in the New text and redundant copies of earlier text from one source SHALL
same packet if available for transmission at the same time. Text be transmitted in the same packet if available for transmission at
from different sources MUST NOT be transmitted in the same packet. the same time. Text from different sources MUST NOT be transmitted
in the same packet.
3.8. Do not send received text to the originating source 3.6. Do not send received text to the originating source
Text received to a mixer from a participant SHOULD NOT be included in Text received by a mixer from a participant SHOULD NOT be included in
transmission from the mixer to that participant. transmission from the mixer to that participant.
3.9. Clean incoming text 3.7. Clean incoming text
A mixer SHALL handle reception, recovery from packet loss, deletion A mixer SHALL handle reception, recovery from packet loss, deletion
of superfluous redundancy, marking of possible text loss and deletion of superfluous redundancy, marking of possible text loss and deletion
of 'BOM' characters from each participant before queueing received of 'BOM' characters from each participant before queueing received
text for transmission to receiving participants. text for transmission to receiving participants.
3.10. Redundant transmission principles 3.8. Redundant transmission principles
A transmitting party using redundancy SHALL send redundant A transmitting party using redundancy SHALL send redundant
repetitions of T140blocks already transmitted in earlier packets. repetitions of T140blocks already transmitted in earlier packets.
The number of redundant generations of T140blocks to include in The number of redundant generations of T140blocks to include in
transmitted packets SHALL be deduced from the SDP negotiation. It transmitted packets SHALL be deduced from the SDP negotiation. It
SHOULD be set to the minimum of the number declared by the two SHOULD be set to the minimum of the number declared by the two
parties negotiating a connection. It is RECOMMENDED to declare and parties negotiating a connection. It is RECOMMENDED to declare and
transmit one original and two redundant generations of the transmit one original and two redundant generations of the
T140blocks. T140blocks.
3.11. Interleaving text from different sources 3.9. Interleaving text from different sources
When text from more than one source is available for transmission When text from more than one source is available for transmission
from a mixer, the mixer SHALL let the sources take turns in having from a mixer, the mixer SHALL let the sources take turns in having
their text transmitted. their text transmitted.
The source with the oldest text received in the mixer or oldest The source with the oldest text received in the mixer or oldest
redundant text SHOULD be next in turn to get all its available unsent redundant text SHOULD be next in turn to get all its available unsent
text transmitted. The age of redundant text SHOULD then be text transmitted. The age of redundant text SHOULD then be
considered to be 330 ms after its previous transmission. considered to be 330 ms after its previous transmission.
3.12. Text placement in packets 3.10. Text placement in packets
The mixer SHOULD compose and transmit an RTP packet to a receiver The mixer SHOULD compose and transmit an RTP packet to a receiver
when one of the following conditions has occurred: when one of the following conditions has occurred:
* 100 ms has passed since the latest transmission to that receiver, * 100 ms has passed since the latest transmission to that receiver,
and there is unsent text available for transmission. and there is unsent text available for transmission.
* New text has arrived and more than 100 ms has passed since latest * New text has arrived and more than 100 ms has passed since latest
transmission to that receiver. transmission to that receiver.
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since the latest transmission to that receiver, and the redundant since the latest transmission to that receiver, and the redundant
text is still not sent. text is still not sent.
At time of transmission, the mixer SHALL populate the RTP packet with At time of transmission, the mixer SHALL populate the RTP packet with
all T140blocks queued for transmission originating from the source in all T140blocks queued for transmission originating from the source in
turn for transmission as long as this is not in conflict with the turn for transmission as long as this is not in conflict with the
allowed number of characters per second ("CPS") or the maximum packet allowed number of characters per second ("CPS") or the maximum packet
size. In this way, the latency of the latest received text is kept size. In this way, the latency of the latest received text is kept
low even in moments of simultaneous transmission from many sources. low even in moments of simultaneous transmission from many sources.
Redundant text SHALL also be included. See Section 3.14 Redundant text SHALL also be included. See Section 3.12
The SSRC of the source shall be placed as the only member in the The SSRC of the source SHALL be placed as the only member in the
CSRC-list. CSRC-list.
Note: The CSRC-list in an RTP packet only includes the participant Note: The CSRC-list in an RTP packet only includes the participant
who's text is included in text blocks. It is not the same as the 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 total list of participants in a conference. With audio and video
media, the CSRC-list would often contain all participants who are not media, the CSRC-list would often contain all participants who are not
muted whereas text participants that don't type are completely silent muted whereas text participants that don't type are completely silent
and thus are not represented in RTP packet CSRC-lists. and thus are not represented in RTP packet CSRC-lists.
3.13. Empty T140blocks 3.11. Empty T140blocks
If no unsent T140blocks were available for a source at the time of If no unsent T140blocks were available for a source at the time of
populating a packet, but T140blocks are available which have not yet populating a packet, but T140blocks are available which have not yet
been sent the full intended number of redundant transmissions, then been sent the full intended number of redundant transmissions, then
the primary T140block for that source is composed of an empty the primary T140block for that source is composed of an empty
T140block, and populated (without taking up any length) in a packet T140block, and populated (without taking up any length) in a packet
for transmission. The corresponding SSRC SHALL be placed as usual in for transmission. The corresponding SSRC SHALL be placed as usual in
its place in the CSRC-list. its place in the CSRC-list.
The first packet in the session, the first after a source switch and The first packet in the session, the first after a source switch and
the first after a pause SHALL be poulated with the available the first after a pause SHALL be poulated with the available
T140blocks for the source in turn to be sent as primary, and empty T140blocks for the source in turn to be sent as primary, and empty
T140blocks for the agreed number of redundancy generations. T140blocks for the agreed number of redundancy generations.
3.14. Creation of the redundancy 3.12. Creation of the redundancy
The primary T140block from a source in the latest transmitted packet The primary T140block from a source in the latest transmitted packet
is saved for populating the first redundant T140block for that source is saved for populating the first redundant T140block for that source
in next transmission of text from that source. The first redundant in next transmission of text from that source. The first redundant
T140block for that source from the latest transmission is saved for T140block for that source from the latest transmission is saved for
populating the second redundant T140block in next transmission of populating the second redundant T140block in next transmission of
text from that source. text from that source.
Usually this is the level of redundancy used. If a higher number of Usually this is the level of redundancy used. If a higher number of
redundancy is negotiated, then the procedure SHALL be maintained redundancy is negotiated, then the procedure SHALL be maintained
until all available redundant levels of T140blocks are placed in the until all available redundant levels of T140blocks are placed in the
packet. If a receiver has negotiated a lower number of "text/red" packet. If a receiver has negotiated a lower number of "text/red"
generations, then that level shall be the maximum used by the generations, then that level SHOULD be the maximum used by the
transmitter. transmitter.
The T140blocks saved for transmission as redundant data are assigned The T140blocks saved for transmission as redundant data are assigned
a planned transmission time 330 ms after the current time, but SHOULD 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 be transmitted earlier if new text for the same source gets in turn
for transmission before that time. for transmission before that time.
3.15. Timer offset fields 3.13. Timer offset fields
The timestamp offset values are inserted in the redundancy header, The timestamp offset values are inserted in the redundancy header,
with the time offset from the RTP timestamp in the packet when the with the time offset from the RTP timestamp in the packet when the
corresponding T140block was sent as primary. corresponding T140block was sent as primary.
The timestamp offsets are expressed in the same clock tick units as The timestamp offsets are expressed in the same clock tick units as
the RTP timestamp. the RTP timestamp.
The timestamp offset values for empty T140blocks have no relevance The timestamp offset values for empty T140blocks have no relevance
but SHOULD be assigned realistic values. but SHOULD be assigned realistic values.
3.16. Other RTP header fields 3.14. Other RTP header fields
The number of members in the CSRC list ( 0 or 1) shall be placed in The number of members in the CSRC list ( 0 or 1) SHALL be placed in
the "CC" header field. Only mixers place value 1 in the "CC" field. the "CC" header field. Only mixers place value 1 in the "CC" field.
A value of "0" indicates that the source is the transmitting device A value of "0" indicates that the source is the transmitting device
itself and that the source is indicated by the SSRC field. This itself and that the source is indicated by the SSRC field. This
value is used by endpoints, and by mixers sending data that it is value is used by endpoints, and by mixers sending data that it is
source of itself. source of itself.
The current time SHALL be inserted in the timestamp. The current time SHALL be inserted in the timestamp.
The SSRC of the mixer for the RTT session SHALL be inserted in the The SSRC of the mixer for the RTT session SHALL be inserted in the
SSRC field of the RTP header. SSRC field of the RTP header.
The M-bit shall be handled as specified in [RFC4103]. The M-bit SHALL be handled as specified in [RFC4103].
3.17. Pause in transmission 3.15. Pause in transmission
When there is no new T140block to transmit, and no redundant When there is no new T140block to transmit, and no redundant
T140block that has not been retransmitted the intended number of T140block that has not been retransmitted the intended number of
times from any source, the transmission process can stop until either times from any source, the transmission process can stop until either
new T140blocks arrive, or a keep-alive method calls for transmission new T140blocks arrive, or a keep-alive method calls for transmission
of keep-alive packets. of keep-alive packets.
3.18. RTCP considerations 3.16. RTCP considerations
A mixer SHALL send RTCP reports with SDES, CNAME and NAME information A mixer SHALL send RTCP reports with SDES, CNAME and NAME information
about the sources in the multi-party call. This makes it possible about the sources in the multi-party call. This makes it possible
for participants to compose a suitable label for text from each for participants to compose a suitable label for text from each
source. source.
Integrity considerations SHALL be considered when composing these Integrity SHALL be considered when composing these fields. They
fields. contain name and address information that may be sensitive to
transmit in its entirety e.g. to unauthenticated participants.
Similar considerations SHOULD be taken as for other media.
3.19. Reception of multi-party contents 3.17. Reception of multi-party contents
The "text/red" receiver included in an endpoint with presentation The "text/red" receiver included in an endpoint with presentation
functions will receive RTP packets in the single stream from the functions will receive RTP packets in the single stream from the
mixer, and SHALL distribute the T140blocks for presentation in mixer, and SHALL distribute the T140blocks for presentation in
presentation areas for each source. Other receiver roles, such as presentation areas for each source. Other receiver roles, such as
gateways or chained mixers are also feasible, and requires gateways or chained mixers are also feasible, and requires
consideration if the stream shall just be forwarded, or distributed consideration if the stream shall just be forwarded, or distributed
based on the different sources. based on the different sources.
3.19.1. Multi-party vs two-party use 3.17.1. Acting on the source of the packet contents
If the "CC" field value of a received packet is 1, it indicates that If the "CC" field value of a received packet is 1, it indicates that
multi-party transmission is active, and the receiver MUST be prepared the text is conveyed from a source indicated in the single member in
to act on the source according to its role. If the CC value is 0, the CSRC-list, and the receiver MUST act on the source according to
the connection is point-to-point. its role. If the CC value is 0, the source is indicated in the SSRC
field.
3.19.2. Level of redundancy
The used level of redundancy generations SHALL be evaluated from the
received packet contents. The number of generations (including the
primary) is equal to the number of members in the redundancy header.
3.19.3. Empty T140blocks
Empty T140blocks are included as fillers for unused primary or
redundancy levels in the packets. They just do not provide any
contents.
3.19.4. Detection and indication of possible text loss 3.17.2. Detection and indication of possible text loss
The RTP sequence numbers of the received packets SHALL be monitored The RTP sequence numbers of the received packets SHALL be monitored
for gaps and packets out of order. If a sequence number gap appears for gaps and packets out of order. If a sequence number gap appears
and still exists after some defined short time for jitter resolution, and still exists after some defined short time for jitter resolution,
the packets in the gap SHALL be regarded lost. the packets in the gap SHALL be regarded as lost.
If it is known that only one source is active in the RTP session, 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 then it is likely that a gap equal to or larger than the agreed
number of redundancy generations (including the primary) causes text number of redundancy generations (including the primary) causes text
loss. In that case a t140block SHALL be created with a marker for loss. In that case a t140block SHALL be created with a marker for
possible text loss [T140ad1] and assigned to the source and inserted possible text loss [T140ad1] and assigned to the source and inserted
in the reception buffer for that source. in the reception buffer for that source.
If it is known that more than one source is active in the RTP 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 session, then it is not possible in general to evaluate if text was
lost when packets were lost. With two active sources and the lost when packets were lost. With two active sources and the
recommended number of redundancy generations (3), it can take a gap recommended number of redundancy generations (3), it can take a gap
of five consecutive lost packets until any text may be lost, but text 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 loss can also appear if three non-consecutive packets are lost when
they contained consecutive data from the same source. A simple they contained consecutive data from the same source. A simple
method to decide when there is risk for resulting text loss is to method to decide when there is risk for resulting text loss is to
evaluate if three or more packets were lost within one second. Then evaluate if three or more packets were lost within one second. Then
a t140block SHALL be created with a marker for possible text loss a t140block SHOULD be created with a marker for possible text loss
[T140ad1] and assigned to the SSRC of the transmitter as a general [T140ad1] and assigned to the SSRC of the transmitter as a general
input from the mixer. input from the mixer.
Implementations MAY apply more refined methods for more reliable Implementations MAY apply more refined methods for more reliable
detection of if text was lost or not. Any refined method SHOULD detection of if text was lost or not. Any refined method SHOULD
rather falsely mark possible loss when there was no loss instead of prefer marking possible loss rather than not marking when it is
not marking possible loss when there was loss. uncertain if there was loss.
3.19.5. Extracting text and handling recovery 3.17.3. Extracting text and handling recovery
When applying the following procedures, the effects MUST be When applying the following procedures, the effects MUST be
considered of possible timestamp wrap around and the RTP session considered of possible timestamp wrap around and the RTP session
possibly changing SSRC. possibly changing SSRC.
When a packet is received in an RTP session using the packetization When a packet is received in an RTP session using the packetization
for multi-party aware endpoints, its T140blocks SHALL be extracted in for multi-party aware endpoints, its T140blocks SHALL be extracted in
the following way. The description is adapted to the default the following way. The description is adapted to the default
redundancy case using the original and two redundant generations. redundancy case using the original and two redundant generations.
skipping to change at page 17, line 19 skipping to change at page 18, line 33
If the packet is not the first packet from a source, then if the If the packet is not the first packet from a source, then if the
second generation redundant data is available, its timestamp SHALL be second generation redundant data is available, its timestamp SHALL be
created by subtracting its timestamp offset from the RTP timestamp. created by subtracting its timestamp offset from the RTP timestamp.
If the resulting timestamp is later than the latest retrieved data If the resulting timestamp is later than the latest retrieved data
from the same source, then the redundant data SHALL be retrieved and from the same source, then the redundant data SHALL be retrieved and
appended to the receive buffer. The process SHALL be continued in appended to the receive buffer. The process SHALL be continued in
the same way for the first generation redundant data. After that, the same way for the first generation redundant data. After that,
the primary data SHALL be retrieved from the packet and appended to the primary data SHALL be retrieved from the packet and appended to
the receive buffer for the source. the receive buffer for the source.
3.19.6. Delete 'BOM' 3.17.4. Delete 'BOM'
Unicode character 'BOM' is used as a start indication and sometimes Unicode character 'BOM' is used as a start indication and sometimes
used as a filler or keep alive by transmission implementations. used as a filler or keep alive by transmission implementations.
These SHALL be deleted after extraction from received packets. These SHALL be deleted after extraction from received packets.
3.20. Performance considerations 3.18. Performance considerations
This solution has good performance for up to five participants This solution has good performance for up to five participants
simultaneously sending text. At higher numbers of participants simultaneously sending text. At higher numbers of participants
simultaneously sending text, a jerkiness is visible in the simultaneously sending text, a jerkiness is visible in the
presentation of text. With ten participants simultaneously presentation of text. With ten participants simultaneously
transmitting text, the jerkiness is about one second. Even so, the transmitting text, the jerkiness is about one second. Even so, the
transmission of text catches up, so there is limited delay of new transmission of text catches up, so there is limited delay of new
text. The solution is therefore suitable for emergency service use, text. The solution is therefore suitable for emergency service use,
relay service use, and small or well-managed larger multimedia relay service use, and small or well-managed larger multimedia
conferences. Only in large unmanaged conferences with a high number conferences. Only in large unmanaged conferences with a high number
of participants there may on very rare occasions appear situations of participants there may on very rare occasions appear situations
when many participants happen to send text simultaneously, resulting when many participants happen to send text simultaneously, resulting
in unpleasantly jerky presentation of text from each sending in unpleasantly jerky presentation of text from each sending
participant. It should be noted that it is only the number of users participant. It should be noted that it is only the number of users
sending text within the same moment that causes jerkiness, not the sending text within the same moment that causes jerkiness, not the
total number of users with RTT capability. total number of users with RTT capability.
3.21. Security for session control and media 3.19. Security for session control and media
Security SHOULD be applied on both session control and media. In Security SHOULD be applied by use of SIP over TLS by default
applications where legacy endpoints without security may exist, a according to [RFC5630] section 3.1.3 on session control level and by
negotiation SHOULD be performed to decide if security by encryption default using DTLS-SRTP [RFC5764] on media level. In applications
will be applied. If no other security solution is mandated for the where legacy endpoints without security may exist, a negotiation
application, then RFC 8643 OSRTP [RFC8643] SHOULD be applied to SHOULD be performed to decide if security by encryption on media
negotiate SRTP media security with DTLS. Most SDP examples below are level will be applied. If no other security solution is mandated for
for simplicity expressed without the security additions. The the application, then OSRTP [RFC8643] is a suitable method be applied
to negotiate SRTP media security with DTLS. Most SDP examples below
are for simplicity expressed without the security additions. The
principles (but not all details) for applying DTLS-SRTP [RFC5764] principles (but not all details) for applying DTLS-SRTP [RFC5764]
security is shown in a couple of the following examples. security is shown in a couple of the following examples.
3.22. SDP offer/answer examples 3.20. SDP offer/answer examples
This sections shows some examples of SDP for session negotiation of This sections shows some examples of SDP for session negotiation of
the real-time text media in SIP sessions. Audio is usually provided the real-time text media in SIP sessions. Audio is usually provided
in the same session, and sometimes also video. The examples only in the same session, and sometimes also video. The examples only
show the part of importance for the real-time text media. show the part of importance for the real-time text media. The
examples relate to the single RTP stream mixing for multi-party aware
endpoints and for multi-party unaware endpoints.
Note: Multi-party RTT may also be provided through other methods,
e.g. by a Selective Forwarding Middlebox (SFM). In that case, the
SDP of the offer will include something specific for that method, and
an answer acknowledging the use of that method would accept it by
something specific included in the SDP. The offer may contain also
the "rtt-mixer" sdp media attribute for the main RTT media when the
offeror has capability for both multi-party methods, while an answer,
selecting to use SFM will not include the "rtt-mixer" sdp media
attribute.
Offer example for "text/red" format and multi-party support: Offer example for "text/red" format and multi-party support:
m=text 11000 RTP/AVP 100 98 m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000 a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000 a=rtpmap:100 red/1000
a=fmtp:100 98/98/98 a=fmtp:100 98/98/98
a=rtt-mixer a=rtt-mixer
Answer example from a multi-party capable device Answer example from a multi-party capable device
m=text 14000 RTP/AVP 100 98 m=text 14000 RTP/AVP 100 98
a=rtpmap:98 t140/1000 a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000 a=rtpmap:100 red/1000
a=fmtp:100 98/98/98 a=fmtp:100 98/98/98
a=rtt-mixer a=rtt-mixer
Offer example for "text/red" format including multi-party Offer example for "text/red" format including multi-party
and security: and security:
a=fingerprint: (fingerprint1) a=fingerprint: (fingerprint1)
m=text 11000 RTP/AVP 100 98 m=text 11000 RTP/AVP 100 98
skipping to change at page 19, line 23 skipping to change at page 20, line 45
With the "fingerprint" the device acknowledges use of SRTP/DTLS. With the "fingerprint" the device acknowledges use of SRTP/DTLS.
Answer example from a multi-party unaware device that also Answer example from a multi-party unaware device that also
does not support security: does not support security:
m=text 12000 RTP/AVP 100 98 m=text 12000 RTP/AVP 100 98
a=rtpmap:98 t140/1000 a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000 a=rtpmap:100 red/1000
a=fmtp:100 98/98/98 a=fmtp:100 98/98/98
3.23. Packet sequence example from interleaved transmission 3.21. Packet sequence example from interleaved transmission
This example shows a symbolic flow of packets from a mixer including This example shows a symbolic flow of packets from a mixer including
loss and recovery. The sequence includes interleaved transmission of loss and recovery. The sequence includes interleaved transmission of
text from two RTT sources A and B. P indicates primary data. R1 is text from two RTT sources A and B. P indicates primary data. R1 is
first redundant generation data and R2 is the second redundant first redundant generation data and R2 is the second redundant
generation data. A1, B1, A2 etc are text chunks (T140blocks) generation data. A1, B1, A2 etc are text chunks (T140blocks)
received from the respective sources and sent on to the receiver by received from the respective sources and sent on to the receiver by
the mixer. X indicates dropped packet between the mixer and a the mixer. X indicates dropped packet between the mixer and a
receiver. The session is assumed to use original and two redundant receiver. The session is assumed to use original and two redundant
generations of RTT. generations of RTT.
skipping to change at page 20, line 25 skipping to change at page 21, line 44
B1 is planned 330 ms after packet 102. B1 is planned 330 ms after packet 102.
X------------------------| X------------------------|
X Seq no 103, Timer=20730| X Seq no 103, Timer=20730|
X CC=1 | X CC=1 |
X CSRC list A | X CSRC list A |
X R2: A2, Offset=630 | X R2: A2, Offset=630 |
X R1: A3, Offset=330 | X R1: A3, Offset=330 |
X P: Empty | X P: Empty |
X------------------------| X------------------------|
Packet 103 is assumed to be dropped in network problems. It Packet 103 is assumed to be lost due to network problems.
contains redundancy for A. Sending A3 as second level It contains redundancy for A. Sending A3 as second level
redundancy is planned for 330 ms after packet 104. redundancy is planned for 330 ms after packet 104.
X------------------------| X------------------------|
X Seq no 104, Timer=20820| X Seq no 104, Timer=20820|
X CC=1 | X CC=1 |
X CSRC list B | X CSRC list B |
X R2: Empty, Offset=600 | X R2: Empty, Offset=600 |
X R1: B1, Offset=300 | X R1: B1, Offset=300 |
X P: B2 | X P: B2 |
X------------------------| X------------------------|
skipping to change at page 22, line 5 skipping to change at page 23, line 19
is the timestamp of packet 101 THAT was received. So B1 does not is the timestamp of packet 101 THAT was received. So B1 does not
need to be retrieved. The first level redundancy in packet 106 has need to be retrieved. The first level redundancy in packet 106 has
offset 340. The timestamp of packet 106 minus 340 is 20820. That is offset 340. The timestamp of packet 106 minus 340 is 20820. That is
later than the latest received packet with source B. Therefore B2 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 retrieved and assigned to the input buffer for source B. No primary
is available in packet 106 is available in packet 106
After this sequence, A3 and B1 and B2 have been received. In this After this sequence, A3 and B1 and B2 have been received. In this
case no text was lost. case no text was lost.
3.24. Maximum character rate "CPS" 3.22. Maximum character rate "CPS"
The default maximum rate of reception of "text/t140" real-time text The default maximum rate of reception of "text/t140" real-time text
is in RFC 4103 [RFC4103] specified to be 30 characters per second. is in RFC 4103 [RFC4103] specified to be 30 characters per second.
The value MAY be modified in the CPS parameter of the FMTP attribute 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 in the media section for the "text/t140" media. A mixer combining
real-time text from a number of sources may occasionally have a real-time text from a number of sources may occasionally have a
higher combined flow of text coming from the sources. Endpoints higher combined flow of text coming from the sources. Endpoints
SHOULD therefore specify a suitable higher value for the CPS SHOULD therefore specify a suitable higher value for the CPS
parameter, corresponding to its real reception capability. A value parameter, corresponding to its real reception capability. A value
for "CPS" of 90 is the default for the "text/t140" stream in the for "CPS" of 90 SHALL be the default for the "text/t140" stream in
"text/red" format when multi-party real-time text is negotiated. See the "text/red" format when multi-party real-time text is negotiated.
RFC 4103 [RFC4103] for the format and use of the CPS parameter. The See RFC 4103 [RFC4103] for the format and use of the CPS parameter.
same rules apply for the multi-party case except for the default The same rules apply for the multi-party case except for the default
value. value.
4. Presentation level considerations 4. Presentation level considerations
ITU-T T.140 [T140] provides the presentation level requirements for ITU-T T.140 [T140] provides the presentation level requirements for
the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for
erasure and other formatting functions and has the following general erasure and other formatting functions and has the following general
statement for the presentation: statement for the presentation:
"The display of text from the members of the conversation should be "The display of text from the members of the conversation should be
skipping to change at page 25, line 9 skipping to change at page 26, line 9
|Eve, will you do | | | |Eve, will you do | | |
|your presentation on| | | |your presentation on| | |
|Friday? |Yes, Friday at 10. | | |Friday? |Yes, Friday at 10. | |
|Fine, wo | |We need to meet befo | |Fine, wo | |We need to meet befo |
|___________________________________________________________________| |___________________________________________________________________|
Figure 4: An example of a coordinated column-view of a three-party Figure 4: An example of a coordinated column-view of a three-party
session with entries ordered vertically in approximate time-order. session with entries ordered vertically in approximate time-order.
4.2. Multi-party mixing for multi-party unaware endpoints 4.2. Multi-party mixing for multi-party unaware endpoints
When the mixer has indicated multi-party capability by the "rtt- When the mixer has indicated RTT multi-party capability in an SDP
mixer" sdp attribute in an SDP negotiation, but the multi-party negotiation, but the multi-party capability negotiation fails with an
capability negotiation fails with an endpoint, then the agreed "text/ endpoint, then the agreed "text/red" or "text/t140" format SHALL be
red" or "text/t140" format SHALL be used and the mixer SHOULD compose used and the mixer SHOULD compose a best-effort presentation of
a best-effort presentation of multi-party real-time text in one multi-party real-time text in one stream intended to be presented by
stream intended to be presented by an endpoint with no multi-party an endpoint with no multi-party awareness.
awareness.
This presentation format has functional limitations and SHOULD be This presentation format has functional limitations and SHOULD be
used only to enable participation in multi-party calls by legacy used only to enable participation in multi-party calls by legacy
deployed endpoints implementing only RFC 4103 without any multi-party deployed endpoints implementing only RFC 4103 without any multi-party
extensions specified in this document. extensions specified in this document.
The principles and procedures below do not specify any new protocol The principles and procedures below do not specify any new protocol
elements. They are instead composed from the information in ITU-T elements. They are instead composed from the information in ITU-T
T.140 [T140] and an ambition to provide a best effort presentation on T.140 [T140] and an ambition to provide a best effort presentation on
an endpoint which has functions only for two-party calls. an endpoint which has functions only for two-party calls.
The mixer mixing for multi-party unaware endpoints SHALL compose a The mixer mixing for multi-party unaware endpoints SHALL compose a
simulated limited multi-party RTT view suitable for presentation in simulated limited multi-party RTT view suitable for presentation in
one presentation area. The mixer SHALL group text in suitable groups one presentation area. The mixer SHALL group text in suitable groups
and prepare for presentation of them by inserting a new line between and prepare for presentation of them by inserting a new line between
them if the transmitted text did not already end with a new line. A them if the transmitted text did not already end with a new line. A
presentable label SHOULD be composed and sent for the source presentable label SHOULD be composed and sent for the source
initially in the session and after each source switch. With this initially in the session and after each source switch. With this
procedure the time for source switching is depending on the actions procedure the time for switching from transmission of text from one
of the users. In order to expedite source switch, a user can for source to transmission of text from another source is depending on
example end its turn with a new line. the actions of the users. In order to expedite source switch, a user
can for example end its turn with a new line.
4.2.1. Actions by the mixer at reception from the call participants 4.2.1. Actions by the mixer at reception from the call participants
When text is received by the mixer from the different 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 mixer SHALL recover text from redundancy if any packets are lost.
The mark for lost text [T140ad1] SHOULD be inserted in the stream if The mark for lost text [T140ad1] SHOULD be inserted in the stream if
unrecoverable loss appears. Any Unicode "BOM" characters, possibly unrecoverable loss appears. Any Unicode "BOM" characters, possibly
used for keep-alive shall be deleted. The time of creation of text used for keep-alive SHALL be deleted. The time of creation of text
(retrieved from the RTP timestamp) SHALL be stored together with the (retrieved from the RTP timestamp) SHOULD be stored together with the
received text from each source in queues for transmission to the received text from each source in queues for transmission to the
recipients. recipients.
4.2.2. Actions by the mixer for transmission to the recipients 4.2.2. Actions by the mixer for transmission to the recipients
The following procedure SHOULD be applied for each recipient of The following procedure SHOULD be applied for each multi-party
multi-part text from the mixer. unaware recipient of multi-party text from the mixer.
The text for transmission SHOULD be formatted by the mixer for each The text for transmission SHOULD be formatted by the mixer for each
receiving user for presentation in one single presentation area. receiving user for presentation in one single presentation area.
Text received from a participant SHOULD NOT be included in Text received from a participant SHOULD NOT be included in
transmission to that participant. When there is text available for transmission to that participant. When there is text available for
transmission from the mixer to a receiving party from more than one transmission from the mixer to a receiving party from more than one
participant, the mixer SHOULD switch between transmission of text participant, the mixer SHOULD switch between transmission of text
from the different sources at suitable points in the transmitted from the different sources at suitable points in the transmitted
stream. stream.
skipping to change at page 27, line 14 skipping to change at page 28, line 14
When switching source, the source which has the oldest text in queue When switching source, the source which has the oldest text in queue
SHOULD be selected to be transmitted. A character display count SHOULD be selected to be transmitted. A character display count
SHOULD be maintained for the currently transmitted source, starting SHOULD be maintained for the currently transmitted source, starting
at zero after the label is transmitted for the currently transmitted at zero after the label is transmitted for the currently transmitted
source. source.
The status SHOULD be maintained for the latest control code for The status SHOULD be maintained for the latest control code for
Select Graphic Rendition (SGR) from each source. If there is an SGR Select Graphic Rendition (SGR) from each source. If there is an SGR
code stored as the status for the current source before the source 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 switch is done, a reset of SGR SHOULD be sent by the sequence SGR 0
[009B 0000 006D] after the new line and before the new label during a [009B 0000 006D] after the new line and before the new label during a
source switch. See SGR below for an explanation. This transmission source switch. See SGR below for an explanation. This transmission
does not influence the display count. does not influence the display count.
If there is an SGR code stored for the new source after the source If there is an SGR code stored for the new source after the source
switch, that SGR code SHOULD be transmitted to the recipient before switch, that SGR code SHOULD be transmitted to the recipient before
the label. This transmission does not influence the display count. the label. This transmission does not influence the display count.
4.2.3. Actions on transmission of text 4.2.3. Actions on transmission of text
skipping to change at page 27, line 36 skipping to change at page 28, line 36
count by one per transmitted character. count by one per transmitted character.
4.2.4. Actions on transmission of control codes 4.2.4. Actions on transmission of control codes
The following control codes specified by T.140 require specific The following control codes specified by T.140 require specific
actions. They SHOULD cause specific considerations in the mixer. actions. They SHOULD cause specific considerations in the mixer.
Note that the codes presented here are expressed in UCS-16, while Note that the codes presented here are expressed in UCS-16, while
transmission is made in UTF-8 transform of these codes. transmission is made in UTF-8 transform of these codes.
BEL 0007 Bell Alert in session, provides for alerting during an BEL 0007 Bell Alert in session, provides for alerting during an
active session. The display count SHOULD not be altered. active session. The display count SHOULD NOT be altered.
NEW LINE 2028 Line separator. Check and perform a source switch if NEW LINE 2028 Line separator. Check and perform a source switch if
appropriate. Increase display count by 1. appropriate. Increase display count by 1.
CR LF 000D 000A A supported, but not preferred way of requesting a CR LF 000D 000A A supported, but not preferred way of requesting a
new line. Check and perform a source switch if appropriate. new line. Check and perform a source switch if appropriate.
Increase display count by 1. Increase display count by 1.
INT ESC 0061 Interrupt (used to initiate mode negotiation INT ESC 0061 Interrupt (used to initiate mode negotiation
procedure). The display count SHOULD not be altered. procedure). The display count SHOULD NOT be altered.
SGR 009B Ps 006D Select graphic rendition. Ps is rendition SGR 009B Ps 006D Select graphic rendition. Ps is rendition
parameters specified in ISO 6429. The display count SHOULD not be parameters specified in ISO 6429. The display count SHOULD NOT be
altered. The SGR code SHOULD be stored for the current source. altered. The SGR code SHOULD be stored for the current source.
SOS 0098 Start of string, used as a general protocol element SOS 0098 Start of string, used as a general protocol element
introducer, followed by a maximum 256 bytes string and the ST. introducer, followed by a maximum 256 bytes string and the ST.
The display count SHOULD not be altered. The display count SHOULD NOT be altered.
ST 009C String terminator, end of SOS string. The display count ST 009C String terminator, end of SOS string. The display count
SHOULD not be altered. SHOULD NOT be altered.
ESC 001B Escape - used in control strings. The display count SHOULD ESC 001B Escape - used in control strings. The display count SHOULD
not be altered for the complete escape code. NOT be altered for the complete escape code.
Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used
for synchronization and keep-alive. SHOULD be deleted from for synchronization and keep-alive. SHOULD be deleted from
incoming streams. Shall be sent first after session establishment incoming streams. Shall be sent first after session establishment
to the recipient. The display count shall not be altered. to the recipient. The display count SHOULD NOT be altered.
Missing text mark (U+FFFD) "Replacement character", represented as a Missing text mark (U+FFFD) "Replacement character", represented as a
question mark in a rhombus, or if that is not feasible, replaced question mark in a rhombus, or if that is not feasible, replaced
by an apostrophe ', marks place in stream of possible text loss. by an apostrophe ', marks place in stream of possible text loss.
SHOULD be inserted by the reception procedure in case of SHOULD be inserted by the reception procedure in case of
unrecoverable loss of packets. The display count SHOULD be unrecoverable loss of packets. The display count SHOULD be
increased by one when sent as for any other character. increased by one when sent as for any other character.
SGR If a control code for selecting graphic rendition (SGR), other SGR If a control code for selecting graphic rendition (SGR), other
than reset of the graphic rendition (SGR 0) is sent to a than reset of the graphic rendition (SGR 0) is sent to a
recipient, that control code shall also be stored as status for recipient, that control code SHOULD also be stored as status for
the source in the storage for SGR status. If a reset graphic the source in the storage for SGR status. If a reset graphic
rendition (SGR 0) originated from a source is sent, then the SGR rendition (SGR 0) originated from a source is sent, then the SGR
status storage for that source shall be cleared. The display status storage for that source SHOULD be cleared. The display
count shall not be increased. count SHOULD NOT be increased.
BS (U+0008) Back Space, intended to erase the last entered character BS (U+0008) Back Space, intended to erase the last entered character
by a source. Erasure by backspace cannot always be performed as by a source. Erasure by backspace cannot always be performed as
the erasing party intended. If an erasing action erases all text 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 up to the end of the leading label after a source switch, then the
mixer must not transmit more backspaces. Instead it is mixer MUST NOT transmit more backspaces. Instead it is
RECOMMENDED that a letter "X" is inserted in the text stream for RECOMMENDED that a letter "X" is inserted in the text stream for
each backspace as an indication of the intent to erase more. A each backspace as an indication of the intent to erase more. A
new line is usually coded by a Line Separator, but the character new line is usually coded by a Line Separator, but the character
combination "CRLF" MAY be used instead. Erasure of a new line is combination "CRLF" MAY be used instead. Erasure of a new line is
in both cases done by just one erasing action (Backspace). If the in both cases done by just one erasing action (Backspace). If the
display count has a positive value it is decreased by one when the display count has a positive value it is decreased by one when the
BS is sent. If the display count is at zero, it is not altered. BS is sent. If the display count is at zero, it is not altered.
4.2.5. Packet transmission 4.2.5. Packet transmission
skipping to change at page 32, line 12 skipping to change at page 33, line 12
More information on gateways to textphones (TTYs) is found in RFC More information on gateways to textphones (TTYs) is found in RFC
5194[RFC5194] 5194[RFC5194]
6.2. Gateway considerations with WebRTC. 6.2. Gateway considerations with WebRTC.
Gateway operation to real-time text in WebRTC may also be required. Gateway operation to real-time text in WebRTC may also be required.
In WebRTC, RTT is specified in In WebRTC, RTT is specified in
[I-D.ietf-mmusic-t140-usage-data-channel]. [I-D.ietf-mmusic-t140-usage-data-channel].
A multi-party bridge may have functionality for communicating by RTT A multi-party bridge may have functionality for communicating by RTT
both in RTP streams with RTT and WebRTC t140 data channels. Other both in RTP streams with RTT and WebRTC T.140 data channels. Other
configurations may consist of a multi-party bridge with either configurations may consist of a multi-party bridge with either
technology for RTT transport and a separate gateway for conversion of technology for RTT transport and a separate gateway for conversion of
the text communication streams between RTP and t140 data channel. the text communication streams between RTP and T.140 data channel.
In WebRTC, it is assumed that for a multi-party session, one t140 In WebRTC, it is assumed that for a multi-party session, one T.140
data channel is established for each source from a gateway or bridge data channel is established for each source from a gateway or bridge
to each participant. Each participant also has a data channel with to each participant. Each participant also has a data channel with
two-way connection with the gateway or bridge. two-way connection with the gateway or bridge.
The t140 channel used both ways is for text from the WebRTC user and The t140 channel used both ways is for text from the WebRTC user and
from the bridge or gateway itself to the WebRTC user. The label from the bridge or gateway itself to the WebRTC user. The label
parameter of this t140 channel is used as NAME field in RTCP to parameter of this t140 channel is used as NAME field in RTCP to
participants on the RTP side. The other t140 channels are only for participants on the RTP side. The other t140 channels are only for
text from other participants to the WebRTC user. text from other participants to the WebRTC user.
When a new participant has entered the session with RTP transport of When a new participant has entered the session with RTP transport of
rtt, a new t140 channel SHOULD be established to WebRTC users with RTT, a new T.140 channel SHOULD be established to WebRTC users with
the label parameter composed from the NAME field in RTCP on the RTP the label parameter composed from the NAME field in RTCP on the RTP
side. side.
When a new participant has entered the multi-party session with RTT When a new participant has entered the multi-party session with RTT
transport in a WebRTC t140 data channel, the new participant SHOULD transport in a WebRTC T.140 data channel, the new participant SHOULD
be announced by a notification to RTP users. The label parameter 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 from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP
side, or other available session information. side, or other available session information.
When a participant on the RTP side disappears, the corresponding
T.140 data channel(s) SHOULD be closed.
When a WebRTC user of T.140 data channels disconnects from the mixer,
the corresponding RTP streams or sources in an RTP-mixed stream
SHOULD be closed.
T.140 data channels MAY be opened and closed by negotiation or
renegotiation of the session or by any other valid means as specified
in section 1 of [I-D.ietf-mmusic-t140-usage-data-channel].
7. Updates to RFC 4103 7. Updates to RFC 4103
This document updates RFC 4103[RFC4103] by introducing an sdp media This document updates RFC 4103[RFC4103] by introducing an sdp media
attribute "rtt-mixer" for negotiation of multi-party mixing attribute "rtt-mixer" for negotiation of multi-party mixing
capability with the [RFC4103] format, and by specifying the rules for capability with the [RFC4103] format, and by specifying the rules for
packets when multi-party capability is negotiated and in use. packets when multi-party capability is negotiated and in use.
8. Congestion considerations 8. Congestion considerations
The congestion considerations and recommended actions from RFC 4103 The congestion considerations and recommended actions from RFC 4103
[RFC4103] are valid also in multi-party situations. [RFC4103] are valid also in multi-party situations.
The first action in case of congestion SHOULD be to temporarily The first action in case of congestion SHOULD be to temporarily
increase the transmission interval up to two seconds. increase the transmission interval up to two seconds.
If the unlikely situation appears that more than 20 participants in a If the very unlikely situation appears that more than 70 participants
conference send text simultaneously, it will take more than 7 seconds in a conference send text simultaneously, it will take more than 7
between presentation of text from each of these participants. More seconds between presentation of text from each of these participants.
time than that can cause confusion in the session. It is therefore More time than that can cause confusion in the session. It is
RECOMMENDED that the mixer discards such text in excess inserts a therefore RECOMMENDED that RTP-mixer based mixer discards such text
general indication of possible text loss [T140ad1] in the session. in excess and inserts a general indication of possible text loss
If the main text contributor is indicated in any way, the mixer MAY [T140ad1] in the session. If the main text contributor is indicated
avoid deleting text from that participant. in any way, the mixer MAY avoid deleting text from that participant.
9. Acknowledgements 9. Acknowledgements
James Hamlin for format and performance aspects. James Hamlin for format and performance aspects.
10. IANA Considerations 10. IANA Considerations
10.1. Registration of the "rtt-mixer" sdp media attribute 10.1. Registration of the "rtt-mixer" sdp media attribute
[RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the [RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
RFC number of this document.] RFC number of this document.]
IANA is asked to register the new sdp attribute "rtt-mixer". IANA is asked to register the new sdp attribute "rtt-mixer".
Contact name: IESG Contact name: IESG
Contact email: iesg@ietf.org Contact email: iesg@ietf.org
Attribute name: rtt-mixer Attribute name: rtt-mixer
Attribute semantics: See RFCXXXX Section 3.1 Attribute semantics: See RFCXXXX Section 2.3
Attribute value: none Attribute value: none
Usage level: media Usage level: media
Purpose: Indicate support by mixer and endpoint of multi-party Purpose: Indicate support by mixer and endpoint of multi-party
mixing for real-time text transmission, using a common RTP-stream mixing for real-time text transmission, using a common RTP-stream
for transmission of text from a number of sources mixed with one 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 source at a time and the source indicated in a single CSRC-list
member. member.
Charset Dependent: no Charset Dependent: no
O/A procedure: See RFCXXXX Section 3.1 O/A procedure: See RFCXXXX Section 2.3
Mux Category: normal Mux Category: normal
Reference: RFCXXXX Reference: RFCXXXX
11. Security Considerations 11. Security Considerations
The RTP-mixer model requires the mixer to be allowed to decrypt, pack The RTP-mixer model requires the mixer to be allowed to decrypt, pack
and encrypt secured text from the conference participants. Therefore and encrypt secured text from the conference participants. Therefore
the mixer needs to be trusted. This is similar to the situation for the mixer needs to be trusted. This is similar to the situation for
skipping to change at page 34, line 28 skipping to change at page 35, line 41
Participants with malicious intentions may appear and e.g. disturb Participants with malicious intentions may appear and e.g. disturb
the multi-party session by a continuous flow of text, or masquerading the multi-party session by a continuous flow of text, or masquerading
as text from other participants. Counteractions should be to require as text from other participants. Counteractions should be to require
secure signaling, media and authentication, and to provide higher secure signaling, media and authentication, and to provide higher
level conference functions e.g. for blocking and expelling level conference functions e.g. for blocking and expelling
participants. participants.
12. Change history 12. Change history
12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11 12.1. 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.
12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11
Timestamps and timestamp offsets added to the packet examples in Timestamps and timestamp offsets added to the packet examples in
section 3.23, and the description corrected. section 3.23, and the description corrected.
A number of minor corrections added in sections 3.10 - 3.23. A number of minor corrections added in sections 3.10 - 3.23.
12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10 12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10
The packet composition was modified for interleaving packets from The packet composition was modified for interleaving packets from
different sources. different sources.
The packet reception was modified for the new interleaving method. The packet reception was modified for the new interleaving method.
The packet sequence examples was adjusted for the new interleaving The packet sequence examples was adjusted for the new interleaving
method. method.
Modifications according to responses to Brian Rosen of 2020-11-03 Modifications according to responses to Brian Rosen of 2020-11-03
12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09 12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09
Changed name on the SDP media attribute to "rtt-mixer" Changed name on the SDP media attribute to "rtt-mixer"
Restructure of section 2 for balance between aware and unaware cases. Restructure of section 2 for balance between aware and unaware cases.
Moved conference control to own section. Moved conference control to own section.
Improved clarification of recovery and loss in the packet sequence Improved clarification of recovery and loss in the packet sequence
example. example.
A number of editorial corrections and improvements. A number of editorial corrections and improvements.
12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08 12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08
Deleted the method requiring a new packet format "text/rex" because Deleted the method requiring a new packet format "text/rex" because
of the longer standardization and implementation period it needs. of the longer standardization and implementation period it needs.
Focus on use of RFC 4103 text/red format with shorter transmission Focus on use of RFC 4103 text/red format with shorter transmission
interval, and source indicated in CSRC. interval, and source indicated in CSRC.
12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07 12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07
Added a method based on the "text/red" format and single source per Added a method based on the "text/red" format and single source per
packet, negotiated by the "rtt-mixer" sdp attribute. packet, negotiated by the "rtt-mixer" sdp attribute.
Added reasoning and recommendation about indication of loss. Added reasoning and recommendation about indication of loss.
The highest number of sources in one packet is 15, not 16. Changed. 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 Added in information on update to RFC 4103 that RFC 4103 explicitly
allows addition of FEC method. The redundancy is a kind of forward allows addition of FEC method. The redundancy is a kind of forward
error correction.. error correction..
12.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06 12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06
Improved definitions list format. Improved definitions list format.
The format of the media subtype parameters is made to match the The format of the media subtype parameters is made to match the
requirements. requirements.
The mapping of media subtype parameters to sdp is included. The mapping of media subtype parameters to sdp is included.
The CPS parameter belongs to the t140 subtype and does not need to be The CPS parameter belongs to the t140 subtype and does not need to be
registered here. registered here.
12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05 12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05
nomenclature and editorial improvements nomenclature and editorial improvements
"this document" used consistently to refer to this document. "this document" used consistently to refer to this document.
12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04 12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04
'Redundancy header' renamed to 'data header'. 'Redundancy header' renamed to 'data header'.
More clarifications added. More clarifications added.
Language and figure number corrections. Language and figure number corrections.
12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03 12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03
Mention possible need to mute and raise hands as for other media. Mention possible need to mute and raise hands as for other media.
---done ---- ---done ----
Make sure that use in two-party calls is also possible and explained. Make sure that use in two-party calls is also possible and explained.
- may need more wording - - may need more wording -
Clarify the RTT is often used together with other media. --done-- 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 Tell that text mixing is N-1. A users own text is not received in
the mix. -done- the mix. -done-
In 3. correct the interval to: A "text/rex" transmitter SHOULD send In 3. correct the interval to: A "text/rex" transmitter SHOULD send
packets distributed in time as long as there is something (new or packets distributed in time as long as there is something (new or
redundant T140blocks) to transmit. The maximum transmission interval redundant T140blocks) to transmit. The maximum transmission interval
SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a 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 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 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 than 150 ms, and also the maximum character rate to the receiver is
skipping to change at page 37, line 4 skipping to change at page 38, line 34
simultaneous sending users and introduced delay 16, 150 vs simultaneous sending users and introduced delay 16, 150 vs
requirements 5 vs 500. -done -- requirements 5 vs 500. -done --
Clarify redundancy level per connection. -done- Clarify redundancy level per connection. -done-
Timestamp also for the last data header. To make it possible for all 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 text to have time offset as for transmission from the source. Make
that header equal to the others. -done- that header equal to the others. -done-
Mixer always use the CSRC list, even for its own BOM. -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 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- arrived) in section 3 in one paragraph or close to each other. -done-
Documents the goal of good performance with low delay for 5 Documents the goal of good performance with low delay for 5
simultaneous typers in the introduction. -done- simultaneous typers in the introduction. -done-
Describe better that only primary text shall be sent on to receivers. Describe better that only primary text shall be sent on to receivers.
Redundancy and loss must be resolved by the mixer. -done- Redundancy and loss must be resolved by the mixer. -done-
12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02 12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02
SDP and better description and visibility of security by OSRTP RFC SDP and better description and visibility of security by OSRTP RFC
8634 needed. 8634 needed.
The description of gatewaying to WebRTC extended. The description of gatewaying to WebRTC extended.
The description of the data header in the packet is improved. The description of the data header in the packet is improved.
12.11. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 12.12. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01
2,5,6 More efficient format "text/rex" introduced and attribute 2,5,6 More efficient format "text/rex" introduced and attribute
a=rtt-mix deleted. a=rtt-mix deleted.
3. Brief about use of OSRTP for security included- More needed. 3. Brief about use of OSRTP for security included- More needed.
4. Brief motivation for the solution and why not rtp-translator is 4. Brief motivation for the solution and why not rtp-translator is
used added to intro. used added to intro.
7. More limitations for the multi-party unaware mixing method 7. More limitations for the multi-party unaware mixing method
inserted. inserted.
8. Updates to RFC 4102 and 4103 more clearly expressed. 8. Updates to RFC 4102 and 4103 more clearly expressed.
9. Gateway to WebRTC started. More needed. 9. Gateway to WebRTC started. More needed.
12.12. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03 12.13. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03
to draft-ietf-avtcore-multi-party-rtt-mix-00 to draft-ietf-avtcore-multi-party-rtt-mix-00
Changed file name 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. Replaced CDATA in IANA registration table with better coding.
Converted to xml2rfc version 3. Converted to xml2rfc version 3.
12.13. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02 12.14. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02
to -03 to -03
Changed company and e-mail of the author. Changed company and e-mail of the author.
Changed title to "RTP-mixer formatting of multi-party Real-time text" Changed title to "RTP-mixer formatting of multi-party Real-time text"
to better match contents. to better match contents.
Check and modification where needed of use of RFC 2119 words SHALL Check and modification where needed of use of RFC 2119 words SHALL
etc. etc.
More about the CC value in sections on transmitters and receivers so More about the CC value in sections on transmitters and receivers so
that 1-to-1 sessions do not use the mixer format. that 1-to-1 sessions do not use the mixer format.
Enhanced section on presentation for multi-party-unaware endpoints Enhanced section on presentation for multi-party-unaware endpoints
A paragraph recommending CPS=150 inserted in the performance section. A paragraph recommending CPS=150 inserted in the performance section.
12.14. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01 12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01
to -02 to -02
In Abstract and 1. Introduction: Introduced wording about regulatory In Abstract and 1. Introduction: Introduced wording about regulatory
requirements. requirements.
In section 5: The transmission interval is decreased to 100 ms when In section 5: The transmission interval is decreased to 100 ms when
there is text from more than one source to transmit. there is text from more than one source to transmit.
In section 11 about SDP negotiation, a SHOULD-requirement is In section 11 about SDP negotiation, a SHOULD-requirement is
introduced that the mixer should make a mix for multi-party unaware introduced that the mixer should make a mix for multi-party unaware
skipping to change at page 39, line 14 skipping to change at page 40, line 45
In chapter 9. "Use with SIP centralized conferencing framework" the In chapter 9. "Use with SIP centralized conferencing framework" the
following note is inserted: Note: The CSRC-list in an RTP packet only following note is inserted: Note: The CSRC-list in an RTP packet only
includes participants who's text is included in one or more text includes participants who's text is included in one or more text
blocks. It is not the same as the list of participants in a blocks. It is not the same as the list of participants in a
conference. With audio and video media, the CSRC-list would often conference. With audio and video media, the CSRC-list would often
contain all participants who are not muted whereas text participants 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 that don't type are completely silent and so don't show up in RTP
packet CSRC-lists. packet CSRC-lists.
12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00 12.16. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00
to -01 to -01
Editorial cleanup. Editorial cleanup.
Changed capability indication from fmtp-parameter to SDP attribute Changed capability indication from fmtp-parameter to SDP attribute
"rtt-mix". "rtt-mix".
Swapped order of redundancy elements in the example to match reality. Swapped order of redundancy elements in the example to match reality.
Increased the SDP negotiation section Increased the SDP negotiation section
skipping to change at page 40, line 13 skipping to change at page 41, line 42
<https://www.rfc-editor.org/info/rfc4102>. <https://www.rfc-editor.org/info/rfc4102>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text [RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>. <https://www.rfc-editor.org/info/rfc4103>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566, Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>. July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[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 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010, DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>. <https://www.rfc-editor.org/info/rfc5764>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011, DOI 10.17487/RFC6263, June 2011,
<https://www.rfc-editor.org/info/rfc6263>. <https://www.rfc-editor.org/info/rfc6263>.
[RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T. [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
Stach, "An Opportunistic Approach for Secure Real-time Thomson, "Session Traversal Utilities for NAT (STUN) Usage
Transport Protocol (OSRTP)", RFC 8643, for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
DOI 10.17487/RFC8643, August 2019, October 2015, <https://www.rfc-editor.org/info/rfc7675>.
<https://www.rfc-editor.org/info/rfc8643>.
[T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for [T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for
multimedia application text conversation", February 1998, multimedia application text conversation", February 1998,
<https://www.itu.int/rec/T-REC-T.140-199802-I/en>. <https://www.itu.int/rec/T-REC-T.140-199802-I/en>.
[T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), [T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000),
Protocol for multimedia application text conversation", Protocol for multimedia application text conversation",
February 2000, February 2000,
<https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>. <https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>.
skipping to change at page 41, line 15 skipping to change at page 42, line 47
[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol [RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents", (SIP) Call Control - Conferencing for User Agents",
BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006, BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006,
<https://www.rfc-editor.org/info/rfc4579>. <https://www.rfc-editor.org/info/rfc4579>.
[RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real- [RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real-
Time Text over IP Using the Session Initiation Protocol Time Text over IP Using the Session Initiation Protocol
(SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008, (SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008,
<https://www.rfc-editor.org/info/rfc5194>. <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>.
Author's Address Author's Address
Gunnar Hellstrom Gunnar Hellstrom
Gunnar Hellstrom Accessible Communication Gunnar Hellstrom Accessible Communication
SE-13670 Vendelso SE-13670 Vendelso
Sweden Sweden
Email: gunnar.hellstrom@ghaccess.se Email: gunnar.hellstrom@ghaccess.se
 End of changes. 152 change blocks. 
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