RTP-mixer formatting of multiparty Real-time text
draft-ietf-avtcore-multi-party-rtt-mix-17
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 9071.
|
|
|---|---|---|---|
| Author | Gunnar Hellstrom | ||
| Last updated | 2021-05-11 (Latest revision 2021-05-07) | ||
| Replaces | draft-hellstrom-avtcore-multi-party-rtt-source | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
GENART Last Call review
(of
-14)
by Peter Yee
Ready w/issues
|
||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Associated WG milestone |
|
||
| Document shepherd | Dr. Bernard D. Aboba | ||
| Shepherd write-up | Show Last changed 2021-02-11 | ||
| IESG | IESG state | Became RFC 9071 (Proposed Standard) | |
| Consensus boilerplate | Yes | ||
| Telechat date |
(None)
Needs 5 more YES or NO OBJECTION positions to pass. |
||
| Responsible AD | Murray Kucherawy | ||
| Send notices to | bernard.aboba@gmail.com | ||
| IANA | IANA review state | IANA OK - Actions Needed | |
| IANA expert review state | Expert Reviews OK | ||
| IANA expert review comments | Reviewer approved registration in version 16. |
draft-ietf-avtcore-multi-party-rtt-mix-17
AVTCore G. Hellstrom
Internet-Draft Gunnar Hellstrom Accessible Communication
Updates: 4103 (if approved) 7 May 2021
Intended status: Standards Track
Expires: 8 November 2021
RTP-mixer formatting of multiparty Real-time text
draft-ietf-avtcore-multi-party-rtt-mix-17
Abstract
This document provides enhancements for RFC 4103 real-time text
mixing suitable for a centralized conference model that enables
source identification and rapidly interleaved transmission of text
from different sources. The intended use is for real-time text
mixers and participant endpoints capable of providing an efficient
presentation or other treatment of a multiparty real-time text
session. The specified mechanism builds on the standard use of the
CSRC list in the RTP packet for source identification. The method
makes use of the same "text/t140" and "text/red" formats as for two-
party sessions.
Solutions using multiple RTP streams in the same RTP session are
briefly mentioned, as they could have some benefits over the RTP-
mixer model. The possibility to implement the solution in a wide
range of existing RTP implementations made the RTP-mixer model be
selected to be fully specified in this document.
A capability exchange is specified so that it can be verified that a
mixer and a participant can handle the multiparty-coded real-time
text stream using the RTP-mixer method. The capability is indicated
by use of an SDP media attribute "rtt-mixer".
The document updates RFC 4103 "RTP Payload for Text Conversation".
A specification of how a mixer can format text for the case when the
endpoint is not multiparty-aware is also provided.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 8 November 2021.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/
license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document. Code Components
extracted from this document must include Simplified BSD License text
as described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6
1.2. Selected solution and considered alternatives . . . . . . 7
1.3. Intended application . . . . . . . . . . . . . . . . . . 9
2. Overview of the two specified solutions and selection of
method . . . . . . . . . . . . . . . . . . . . . . . . . 10
2.1. The RTP-mixer-based solution for multiparty-aware
endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10
2.2. Mixing for multiparty-unaware endpoints . . . . . . . . . 11
2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11
2.4. Actions depending on capability negotiation result . . . 13
3. Details for the RTP-mixer-based mixing method for
multiparty-aware endpoints . . . . . . . . . . . . . . . 13
3.1. Use of fields in the RTP packets . . . . . . . . . . . . 13
3.2. Initial transmission of a BOM character . . . . . . . . . 14
3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 14
3.4. Transmission interval . . . . . . . . . . . . . . . . . . 14
3.5. Only one source per packet . . . . . . . . . . . . . . . 15
3.6. Do not send received text to the originating source . . . 15
3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 15
3.8. Redundant transmission principles . . . . . . . . . . . . 15
3.9. Interleaving text from different sources . . . . . . . . 15
3.10. Text placement in packets . . . . . . . . . . . . . . . . 16
3.11. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 16
3.12. Creation of the redundancy . . . . . . . . . . . . . . . 17
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3.13. Timer offset fields . . . . . . . . . . . . . . . . . . . 17
3.14. Other RTP header fields . . . . . . . . . . . . . . . . . 17
3.15. Pause in transmission . . . . . . . . . . . . . . . . . . 18
3.16. RTCP considerations . . . . . . . . . . . . . . . . . . . 18
3.17. Reception of multiparty contents . . . . . . . . . . . . 18
3.18. Performance considerations . . . . . . . . . . . . . . . 20
3.19. Security for session control and media . . . . . . . . . 20
3.20. SDP offer/answer examples . . . . . . . . . . . . . . . . 21
3.21. Packet sequence example from interleaved transmission . . 22
3.22. Maximum character rate "CPS" . . . . . . . . . . . . . . 25
4. Presentation level considerations . . . . . . . . . . . . . . 25
4.1. Presentation by multiparty-aware endpoints . . . . . . . 26
4.2. multiparty mixing for multiparty-unaware endpoints . . . 28
5. Relation to Conference Control . . . . . . . . . . . . . . . 34
5.1. Use with SIP centralized conferencing framework . . . . . 34
5.2. Conference control . . . . . . . . . . . . . . . . . . . 35
6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 35
6.1. Gateway considerations with Textphones . . . . . . . . . 35
6.2. Gateway considerations with WebRTC . . . . . . . . . . . 35
7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 36
8. Congestion considerations . . . . . . . . . . . . . . . . . . 36
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 37
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37
10.1. Registration of the "rtt-mixer" SDP media attribute . . 37
11. Security Considerations . . . . . . . . . . . . . . . . . . . 38
12. Change history . . . . . . . . . . . . . . . . . . . . . . . 38
12.1. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-17 . . . . . . . 38
12.2. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-16 . . . . . . . 38
12.3. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . . 38
12.4. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 39
12.5. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 39
12.6. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 39
12.7. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 40
12.8. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 40
12.9. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 40
12.10. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 40
12.11. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 41
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12.12. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 41
12.13. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 41
12.14. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 41
12.15. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 41
12.16. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 43
12.17. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 43
12.18. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-03 to
draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 43
12.19. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-02 to
-03 . . . . . . . . . . . . . . . . . . . . . . . . . . 43
12.20. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-01 to
-02 . . . . . . . . . . . . . . . . . . . . . . . . . . 44
12.21. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-00 to
-01 . . . . . . . . . . . . . . . . . . . . . . . . . . 45
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 45
13.1. Normative References . . . . . . . . . . . . . . . . . . 45
13.2. Informative References . . . . . . . . . . . . . . . . . 46
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 47
1. Introduction
"RTP Payload for Text Conversation" [RFC4103] specifies use of RTP
[RFC3550] for transmission of real-time text (RTT) and the "text/
t140" format. It also specifies a redundancy format "text/red" for
increased robustness. The "text/red" format is registered in
[RFC4102].
Real-time text is usually provided together with audio and sometimes
with video in conversational sessions.
A requirement related to multiparty sessions from the presentation
level standard T.140 [T140] for real-time text is: "The display of
text from the members of the conversation should be arranged so that
the text from each participant is clearly readable, and its source
and the relative timing of entered text is visualized in the
display."
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Another requirement is that the mixing procedure must not introduce
delays in the text streams that are experienced to be disturbing the
real-time experience of the receiving users.
Use of RTT is increasing, and specifically, use in emergency calls is
increasing. Emergency call use requires multiparty mixing. RFC 4103
"RTP Payload for Text Conversation" mixer implementations can use
traditional RTP functions for source identification, but the
performance of the mixer when giving turns for the different sources
to transmit is limited when using the default transmission
characteristics with redundancy.
The redundancy scheme of [RFC4103] enables efficient transmission of
earlier transmitted redundant text in packets together with new text.
However the redundancy header format has no source indicators for the
redundant transmissions. The redundant parts in a packet must
therefore be from the same source as the new text. The recommended
transmission is one new and two redundant generations of text
(T140blocks) in each packet and the recommended transmission interval
for two-party use is 300 ms.
Real-time text mixers for multiparty sessions need to include the
source with each transmitted group of text from a conference
participant so that the text can be transmitted interleaved with text
groups from different sources at the rate they are created. This
enables the text groups to be presented by endpoints in suitable
grouping with other text from the same source.
The presentation can then be arranged so that text from different
sources can be presented in real-time and easily read. At the same
time it is possible for a reading user to perceive approximately when
the text was created in real time by the different parties. The
transmission and mixing is intended to be done in a general way so
that presentation can be arranged in a layout decided by the
endpoint.
There are existing implementations of RFC 4103 in endpoints without
the updates from this document. These will not be able to receive
and present real-time text mixed for multiparty-aware endpoints.
A negotiation mechanism is therefore needed for verification if the
parties are able to handle a common method for multiparty
transmission and agreeing on using that method.
A fallback mixing procedure is also needed for cases when the
negotiation result indicates that a receiving endpoint is not capable
of handling the mixed format. Multiparty-unaware endpoints would
possibly otherwise present all received multiparty mixed text as if
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it came from the same source regardless of any accompanying source
indication coded in fields in the packet. Or they may have other
undesirable ways of acting on the multiparty content. The fallback
method is called the mixing procedure for multiparty-unaware
endpoints. The fallback method is naturally not expected to meet all
performance requirements placed on the mixing procedure for
multiparty-aware endpoints.
The document updates [RFC4103] by introducing an attribute for
indicating capability for the RTP-mixer-based multiparty mixing case
and rules for source indications and interleaving of text from
different sources.
1.1. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown above.
The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP-
mixer, RTP-translator are defined in [RFC3550].
The term "T140block" is defined in [RFC4103] to contain one or more
T.140 code elements.
"TTY" stands for a textphone type used in North America.
"WebRTC" stands for web-based real-time communication specified by
W3C and IETF. See [RFC8825].
"DTLS-SRTP" is a DTLS extension for use with SRTP/SRTCP specified in
[RFC5764].
"multiparty-aware" describes an endpoint receiving real-time text
from multiple sources through a common conference mixer being able to
present the text in real-time, separated by source, and presented so
that a user can get an impression of the approximate relative timing
of text from different parties.
"multiparty-unaware" describes an endpoint not itself being able to
separate text from different sources when received through a common
conference mixer.
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1.2. Selected solution and considered alternatives
A number of alternatives were considered when searching an efficient
and easily implemented multiparty method for real-time text. This
section explains a few of them briefly.
Multiple RTP streams, one per participant
One RTP stream per source would be sent in the same RTP session
with the "text/red" format. From some points of view, use of
multiple RTP streams, one for each source, sent in the same RTP
session would be efficient, and would use exactly the same packet
format as [RFC4103] and the same payload type. A couple of
relevant scenarios using multiple RTP-streams are specified in
"RTP Topologies" [RFC7667]. One possibility of special interest
is the Selective Forwarding Middlebox (SFM) topology specified in
RFC 7667 section 3.7 that could enable end-to-end encryption. In
contrast to audio and video, real-time text is only transmitted
when the users actually transmit information. Thus an SFM
solution would not need to exclude any party from transmission
under normal conditions. In order to allow the mixer to convey
the packets with the payload preserved and encrypted, an SFM
solution would need to act on some specific characteristics of the
"text/red" format. The redundancy headers are part of the
payload, so the receiver would need to just assume that the
payload type number in the redundancy header is for "text/t140".
The characters per second parameter (CPS) would need to act per
stream. The relation between the SSRC and the source would need
to be conveyed in some specified way, e.g., in the CSRC. Recovery
and loss detection would preferably be based on sequence number
gap detection. Thus sequence number gaps in the incoming stream
to the mixer would need to be reflected in the stream to the
participant, with no new gaps created by the mixer. However, the
RTP implementation in both mixers and endpoints need to support
multiple streams in the same RTP session in order to use this
mechanism. For best deployment opportunity, it should be possible
to upgrade existing endpoint solutions to be multiparty-aware with
a reasonable effort. There is currently a lack of support for
multi-stream RTP in certain implementations. This fact led to
this solution being only briefly mentioned in this document as an
option for further study.
RTP-mixer-based method for multiparty-aware endpoints
The "text/red" format in RFC 4103 is sent with a shorter
transmission interval with the RTP-mixer method and indicating the
source in the CSRC field. The "text/red" format with a "text/
t140" payload in a single RTP stream can be sent when text is
available from the call participants instead of at the regular 300
ms. Transmission of packets with text from different sources can
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then be done smoothly while simultaneous transmission occurs as
long as it is not limited by the maximum character rate "CPS".
With ten participants sending text simultaneously, the switching
and transmission performance is good. With more simultaneously
sending participants, and with receivers having the default
capacity there will be a noticeable jerkiness and delay in text
presentation. The jerkiness will be more expressed the more
participants who send text simultaneously. Two seconds jerkiness
will be noticeable and slightly unpleasant, but it corresponds in
time to what typing humans often cause by hesitation or changing
position while typing. A benefit of this method is that no new
packet format needs to be introduced and implemented. Since
simultaneous typing by more than two parties is very rare, this
method can be used successfully with good performance. Recovery
of text in case of packet loss is based on analysis of timestamps
of received redundancy versus earlier received text. Negotiation
is based on a new SDP media attribute "rtt-mixer". This method is
selected to be the main one specified in this document.
Multiple sources per packet
A new "text" media subtype would be specified with up to 15
sources in each packet. The mechanism would make use of the RTP
mixer model specified in RTP [RFC3550]. Text from up to 15
sources can be included in each packet. Packets are normally sent
with 300 ms intervals. The mean delay will be 150 ms. The
sources are indicated in strict order in the CSRC list of the RTP
packets. A new redundancy packet format is specified. This
method would result in good performance, but would require
standardisation and implementation of new releases in the target
technologies that would take more time than desirable to complete.
It was therefore not selected to be included in this document.
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Mixing for multiparty-unaware endpoints
Presentation of text from multiple parties is prepared by the
mixer in one single stream. It is desirable to have a method that
does not require any modifications in existing user devices
implementing RFC 4103 for RTT without explicit support of
multiparty sessions. This is possible by having the mixer insert
a new line and a text formatted source label before each switch of
text source in the stream. Switch of source can only be done in
places in the text where it does not disturb the perception of the
contents. Text from only one source can be presented in real time
at a time. The delay will therefore vary. The method also has
other limitations, but is included in this document as a fallback
method. In calls where parties take turns properly by ending
their entries with a new line, the limitations will have limited
influence on the user experience. when only two parties send text,
these two will see the text in real time with no delay. This
method is specified as a fallback method in this document.
RTT transport in WebRTC
Transport of real-time text in the WebRTC technology is specified
to use the WebRTC data channel in [RFC8865]. That specification
contains a section briefly describing its use in multiparty
sessions. The focus of this document is RTP transport.
Therefore, even if the WebRTC transport provides good multiparty
performance, it is just mentioned in this document in relation to
providing gateways with multiparty capabilities between RTP and
WebRTC technologies.
1.3. Intended application
The method for multiparty real-time text specified in this document
is primarily intended for use in transmission between mixers and
endpoints in centralised mixing configurations. It is also
applicable between mixers. An often mentioned application is for
emergency service calls with real-time text and voice, where a call
taker wants to make an attended handover of a call to another agent,
and stay to observe the session. Multimedia conference sessions with
support for participants to contribute in text is another
application. Conferences with central support for speech-to-text
conversion is yet another mentioned application.
In all these applications, normally only one participant at a time
will send long text utterances. In some cases, one other participant
will occasionally contribute with a longer comment simultaneously.
That may also happen in some rare cases when text is interpreted to
text in another language in a conference. Apart from these cases,
other participants are only expected to contribute with very brief
utterings while others are sending text.
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Users expect that the text they send is presented in real-time in a
readable way to the other participants even if they send
simultaneously with other users and even when they make brief edit
operations of their text by backspacing and correcting their text.
Text is supposed to be human generated, by some text input means,
such as typing on a keyboard or using speech-to-text technology.
Occasional small cut-and-paste operations may appear even if that is
not the initial purpose of real-time text.
The real-time characteristics of real-time text is essential for the
participants to be able to contribute to a conversation. If the text
is too much delayed from typing a letter to its presentation, then,
in some conference situations, the opportunity to comment will be
gone and someone else will grab the turn. A delay of more than one
second in such situations is an obstacle for good conversation.
2. Overview of the two specified solutions and selection of method
This section contains a brief introduction of the two methods
specified in this document.
2.1. The RTP-mixer-based solution for multiparty-aware endpoints
This method specifies negotiated use of the RFC 4103 format for
multiparty transmission in a single RTP stream. The main purpose of
this document is to specify a method for true multiparty real-time
text mixing for multiparty-aware endpoints that can be widely
deployed. The RTP-mixer-based method makes use of the current format
for real-time text in [RFC4103]. It is an update of RFC 4103 by a
clarification on one way to use it in the multiparty situation. That
is done by completing a negotiation for this kind of multiparty
capability and by interleaving packets from different sources. The
source is indicated in the CSRC element in the RTP packets. Specific
considerations are made to be able to recover text after packet loss.
The detailed procedures for the RTP-mixer-based multiparty-aware case
are specified in Section 3.
Please use [RFC4103] as reference when reading the specification.
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2.2. Mixing for multiparty-unaware endpoints
A method is also specified in this document for cases when the
endpoint participating in a multiparty call does not itself implement
any solution, or not the same, as the mixer. The method requires the
mixer to insert text dividers and readable labels and only send text
from one source at a time until a suitable point appears for source
change. This solution is a fallback method with functional
limitations. It acts on the presentation level.
A party acting as a mixer, which has not negotiated any method for
true multiparty RTT handling, but negotiated a "text/red" or "text/
t140" format in a session with a participant SHOULD in order to
maintain interoperability, if nothing else is specified for the
application, format transmitted text to that participant to be
suitable to present on a multiparty-unaware endpoint as further
specified in Section 4.2.
2.3. Offer/answer considerations
RTP Payload for Text Conversation [RFC4103] specifies use of RTP
[RFC3550], and a redundancy format "text/red" for increased
robustness of real-time text transmission. This document updates
[RFC4103] by introducing a capability negotiation for handling
multiparty real-time text, a way to indicate the source of
transmitted text, and rules for efficient timing of the transmissions
interleaved from different sources.
The capability negotiation for the "RTP-mixer-based multiparty
method" is based on use of the SDP media attribute "rtt-mixer".
The syntax is as follows:
"a=rtt-mixer"
If any other method for RTP-based multiparty real-time text gets
specified by additional work, it is assumed that it will be
recognized by some specific SDP feature exchange.
2.3.1. Initial offer
A party intending to set up a session and being willing to use the
RTP-mixer-based method of this specification for sending or receiving
or both sending and receiving real-time text SHALL include the "rtt-
mixer" SDP attribute in the corresponding "text" media section in the
initial offer.
The party MAY indicate capability for both the RTP-mixer-based method
of this specification and other methods.
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When the offeror has sent the offer including the "rtt-mixer"
attribute, it MUST be prepared to receive and handle real-time text
formatted according to both the method for multiparty-aware parties
specified in Section 3 in this specification and two-party formatted
real-time text.
2.3.2. Answering the offer
A party receiving an offer containing the "rtt-mixer" SDP attribute
and being willing to use the RTP-mixer-based method of this
specification for sending or receiving or both sending and receiving
SHALL include the "rtt-mixer" SDP attribute in the corresponding
"text" media section in the answer.
If the offer did not contain the "rtt-mixer" attribute, the answer
MUST NOT contain the "rtt-mixer" attribute.
An answer MUST NOT include acceptance of more than one method for
multiparty real-time text in the same RTP session.
When the answer including acceptance is transmitted, the answerer
MUST be prepared to act on received text in the negotiated session
according to the method for multiparty-aware parties specified in
Section 3 of this specification. Reception of text for a two-party
session SHALL also be supported.
2.3.3. Offeror processing the answer
When the answer is processed by the offeror, it MUST act as specified
in Section 2.4
2.3.4. Modifying a session
A session MAY be modified at any time by any party offering a
modified SDP with or without the "rtt-mixer" SDP attribute expressing
a desired change in the support of multiparty real-time text.
If the modified offer adds indication of support for multiparty real-
time text by including the "rtt-mixer" SDP attribute, the procedures
specified in the previous subsections SHALL be applied.
If the modified offer deletes indication of support for multiparty
real-time text by excluding the "rtt-mixer" SDP attribute, the answer
MUST NOT contain the "rtt-mixer" attribute, and both parties SHALL
after processing the SDP exchange NOT send real-time text formatted
for multiparty-aware parties according to this specification.
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2.4. Actions depending on capability negotiation result
A transmitting party SHALL send text according to the RTP-mixer-based
multiparty method only when the negotiation for that method was
successful and when it conveys text for another source. In all other
cases, the packets SHALL be populated and interpreted as for a two-
party session.
A party which has negotiated the "rtt-mixer" SDP media attribute MUST
populate the CSRC-list, and format the packets according to Section 3
if it acts as an rtp-mixer and sends multiparty text.
A party which has negotiated the "rtt-mixer" SDP media attribute MUST
interpret the contents of the "CC" field, the CSRC-list and the
packets according to Section 3 in received RTP packets in the
corresponding RTP stream.
A party which has not successfully completed the negotiation of the
"rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved
from different sources in the same RTP stream as specified in
Section 3. If the party is a mixer and did declare the "rtt-mixer"
SDP media attribute, it SHOULD perform the procedure for multiparty-
unaware endpoints. If the party is not a mixer, it SHOULD transmit
as in a two-party session according to [RFC4103].
3. Details for the RTP-mixer-based mixing method for multiparty-aware
endpoints
3.1. Use of fields in the RTP packets
The CC field SHALL show the number of members in the CSRC list, which
SHALL be one (1) in transmissions from a mixer when conveying text
from other sources in a multiparty session, and otherwise 0.
When text is conveyed by a mixer during a multiparty session, a CSRC
list SHALL be included in the packet. The single member in the CSRC-
list SHALL contain the SSRC of the source of the T140blocks in the
packet.
When redundancy is used, the RECOMMENDED level of redundancy is to
use one primary and two redundant generations of T140blocks. In some
cases, a primary or redundant T140block is empty, but is still
represented by a member in the redundancy header.
In other regards, the contents of the RTP packets are equal to what
is specified in [RFC4103].
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3.2. Initial transmission of a BOM character
As soon as a participant is known to participate in a session with
another entity and is available for text reception, a Unicode BOM
character SHALL be sent to it by the other entity according to the
procedures in this section. If the transmitter is a mixer, then the
source of this character SHALL be indicated to be the mixer itself.
Note that the BOM character SHALL be transmitted with the same
redundancy procedures as any other text.
3.3. Keep-alive
After that, the transmitter SHALL send keep-alive traffic to the
receiver(s) at regular intervals when no other traffic has occurred
during that interval, if that is decided for the actual connection.
It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The
consent check of [RFC7675] is a possible alternative if it is used
anyway for other reasons.
3.4. Transmission interval
A "text/red" or "text/t140" transmitter in a mixer SHALL send packets
distributed in time as long as there is something (new or redundant
T140blocks) to transmit. The maximum transmission interval SHALL
then be 330 ms, when no other limitations cause a longer interval to
be temporarily used. It is RECOMMENDED to send the next packet to a
receiver as soon as new text to that receiver is available, as long
as the maximum character rate ("CPS") to the receiver is not exceeded
during any 10-second interval. The intention of these time intervals
is to keep the latency low and network load limited while keeping
good protection against text loss in bursty packet loss conditions.
The main purpose of the 330 ms interval is for timing of redundant
transmission, when no new text from the same source is available.
If the "CPS" value is reached, longer transmission intervals SHALL be
applied and only as much of the text queued for transmission SHALL be
sent at the end of each transmission interval that can be allowed
without exceeding the "CPS" value, until the transmission rate falls
under the "CPS" value again. Division of text for partial
transmission MUST then be made at T140block borders. See also
Section 8
For a transmitter not acting as a mixer, the transmission interval
principles from [RFC4103] apply, and the transmission interval SHALL
be 300 ms.
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3.5. Only one source per packet
New text and redundant copies of earlier text from one source SHALL
be transmitted in the same packet if available for transmission at
the same time. Text from different sources MUST NOT be transmitted
in the same packet.
3.6. Do not send received text to the originating source
Text received by a mixer from a participant SHOULD NOT be included in
transmission from the mixer to that participant, because the normal
behavior of the endpoint is to present locally-produced text locally.
3.7. Clean incoming text
A mixer SHALL handle reception, recovery from packet loss, deletion
of superfluous redundancy, marking of possible text loss and deletion
of 'BOM' characters from each participant before queueing received
text for transmission to receiving participants.
3.8. Redundant transmission principles
A transmitting party using redundancy SHALL send redundant
repetitions of T140blocks already transmitted in earlier packets.
The number of redundant generations of T140blocks to include in
transmitted packets SHALL be deduced from the SDP negotiation. It
SHALL be set to the minimum of the number declared by the two parties
negotiating a connection. It is RECOMMENDED to declare and transmit
one original and two redundant generations of the T140blocks because
that provides good protection against text loss in case of packet
loss, and low overhead.
3.9. Interleaving text from different sources
When text from more than one source is available for transmission
from a mixer, the mixer SHALL let the sources take turns in having
their text transmitted.
The source with the oldest text received in the mixer or oldest
redundant text SHALL be next in turn to get all its available unsent
text transmitted. Any redundant repetitions of earlier transmitted
text not yet sent the intended number of times SHALL be included as
redundant retransmission in the transmission.
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3.10. Text placement in packets
The mixer SHALL compose and transmit an RTP packet to a receiver when
one of the following conditions has occurred:
* There is unsent text available for transmission to that receiver.
* 330 ms has passed since already transmitted text was queued for
transmission as redundant text.
At the time of transmission, the mixer SHALL populate the RTP packet
with all T140blocks queued for transmission originating from the
source in turn for transmission as long as this is not in conflict
with the allowed number of characters per second ("CPS") or the
maximum packet size. In this way, the latency of the latest received
text is kept low even in moments of simultaneous transmission from
many sources.
Redundant text SHALL also be included, and the assessment of how much
new text can be included within the maximum packet size MUST take
into account that the redundancy has priority to be transmitted in
its entirety. See Section 3.12
The SSRC of the source SHALL be placed as the only member in the
CSRC-list.
Note: The CSRC-list in an RTP packet only includes the participant
whose text is included in text blocks. It is not the same as the
total list of participants in a conference. With audio and video
media, the CSRC-list would often contain all participants who are not
muted whereas text participants that don't type are completely silent
and thus are not represented in RTP packet CSRC-lists.
3.11. Empty T140blocks
If no unsent T140blocks were available for a source at the time of
populating a packet, but T140blocks are available which have not yet
been sent the full intended number of redundant transmissions, then
the primary T140block for that source is composed of an empty
T140block, and populated (without taking up any length) in a packet
for transmission. The corresponding SSRC SHALL be placed as usual in
its place in the CSRC-list.
The first packet in the session, the first after a source switch, and
the first after a pause SHALL be populated with the available
T140blocks for the source in turn to be sent as primary, and empty
T140blocks for the agreed number of redundancy generations.
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3.12. Creation of the redundancy
The primary T140block from a source in the latest transmitted packet
is saved for populating the first redundant T140block for that source
in the next transmission of text from that source. The first
redundant T140block for that source from the latest transmission is
saved for populating the second redundant T140block in the next
transmission of text from that source.
Usually this is the level of redundancy used. If a higher level of
redundancy is negotiated, then the procedure SHALL be maintained
until all available redundant levels of T140blocks are placed in the
packet. If a receiver has negotiated a lower number of "text/red"
generations, then that level SHALL be the maximum used by the
transmitter.
The T140blocks saved for transmission as redundant data are assigned
a planned transmission time 330 ms after the current time, but SHOULD
be transmitted earlier if new text for the same source gets in turn
for transmission before that time.
3.13. Timer offset fields
The timestamp offset values SHALL be inserted in the redundancy
header, with the time offset from the RTP timestamp in the packet
when the corresponding T140block was sent as primary.
The timestamp offsets are expressed in the same clock tick units as
the RTP timestamp.
The timestamp offset values for empty T140blocks have no relevance
but SHOULD be assigned realistic values.
3.14. Other RTP header fields
The number of members in the CSRC list (0 or 1) SHALL be placed in
the "CC" header field. Only mixers place value 1 in the "CC" field.
A value of "0" indicates that the source is the transmitting device
itself and that the source is indicated by the SSRC field. This
value is used by endpoints, and by mixers sending self-sourced data.
The current time SHALL be inserted in the timestamp.
The SSRC of the mixer for the RTT session SHALL be inserted in the
SSRC field of the RTP header.
The M-bit SHALL be handled as specified in [RFC4103].
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3.15. Pause in transmission
When there is no new T140block to transmit, and no redundant
T140block that has not been retransmitted the intended number of
times from any source, the transmission process SHALL be stopped
until either new T140blocks arrive, or a keep-alive method calls for
transmission of keep-alive packets.
3.16. RTCP considerations
A mixer SHALL send RTCP reports with SDES, CNAME, and NAME
information about the sources in the multiparty call. This makes it
possible for participants to compose a suitable label for text from
each source.
Integrity SHALL be considered when composing these fields. They
contain name and address information that may be sensitive to
transmit in its entirety e.g., to unauthenticated participants.
Similar considerations SHALL be taken as for other media.
3.17. Reception of multiparty contents
The "text/red" receiver included in an endpoint with presentation
functions will receive RTP packets in the single stream from the
mixer, and SHALL distribute the T140blocks for presentation in
presentation areas for each source. Other receiver roles, such as
gateways or chained mixers are also feasible, and requires
consideration if the stream shall just be forwarded, or distributed
based on the different sources.
3.17.1. Acting on the source of the packet contents
If the "CC" field value of a received packet is 1, it indicates that
the text is conveyed from a source indicated in the single member in
the CSRC-list, and the receiver MUST act on the source according to
its role. If the CC value is 0, the source is indicated in the SSRC
field.
3.17.2. Detection and indication of possible text loss
The receiver SHALL monitor the RTP sequence numbers of the received
packets for gaps and packets out of order. If a sequence number gap
appears and still exists after some defined short time for jitter
resolution, the packets in the gap SHALL be regarded as lost.
If it is known that only one source is active in the RTP session,
then it is likely that a gap equal to or larger than the agreed
number of redundancy generations (including the primary) causes text
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loss. In that case, the receiver SHALL create a t140block with a
marker for possible text loss [T140ad1] and associate it with the
source and insert it in the reception buffer for that source.
If it is known that more than one source is active in the RTP
session, then it is not possible in general to evaluate if text was
lost when packets were lost. With two active sources and the
recommended number of redundancy generations (3), it can take a gap
of five consecutive lost packets until any text may be lost, but text
loss can also appear if three non-consecutive packets are lost when
they contained consecutive data from the same source. A simple
method to decide when there is risk for resulting text loss is to
evaluate if three or more packets were lost within one second. If
this simple method is used, then a t140block SHOULD be created with a
marker for possible text loss [T140ad1] and associated with the SSRC
of the transmitter as a general input from the mixer.
Implementations MAY apply more refined methods for more reliable
detection of whether text was lost or not. Any refined method SHOULD
prefer marking possible loss rather than not marking when it is
uncertain if there was loss.
3.17.3. Extracting text and handling recovery
When applying the following procedures, the effects MUST be
considered of possible timestamp wrap around and the RTP session
possibly changing SSRC.
When a packet is received in an RTP session using the packetization
for multiparty-aware endpoints, its T140blocks SHALL be extracted in
the following way. The description is adapted to the default
redundancy case using the original and two redundant generations.
The source SHALL be extracted from the CSRC-list if available,
otherwise from the SSRC.
If the received packet is the first packet received from the source,
then all T140blocks in the packet SHALL be retrieved and assigned to
a receive buffer for the source beginning with the second generation
redundancy, continuing with the first generation redundancy and
finally the primary.
Note: The normal case is that in the first packet, only the primary
data has contents. The redundant data has contents in the first
received packet from a source only after initial packet loss.
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If the packet is not the first packet from a source, then if the
second generation redundant data is available, its timestamp SHALL be
created by subtracting its timestamp offset from the RTP timestamp.
If the resulting timestamp is later than the latest retrieved data
from the same source, then the redundant data SHALL be retrieved and
appended to the receive buffer. The process SHALL be continued in
the same way for the first generation redundant data. After that,
the primary data SHALL be retrieved from the packet and appended to
the receive buffer for the source.
3.17.4. Delete 'BOM'
Unicode character 'BOM' is used as a start indication and sometimes
used as a filler or keep alive by transmission implementations.
These SHALL be deleted after extraction from received packets.
3.18. Performance considerations
This solution has good performance with low text delays as long as
the sum of characters per second during any 10-second interval sent
from a number of simultaneously sending participants to a receiving
participant does not reach the 'CPS' value. At higher numbers of
characters per second sent, a jerkiness is visible in the
presentation of text. The solution is therefore suitable for
emergency service use, relay service use, and small or well-managed
larger multimedia conferences. Only in large unmanaged conferences
with a high number of participants there may on very rare occasions
appear situations when many participants happen to send text
simultaneously, resulting in unpleasantly jerky presentation of text
from each sending participant. It should be noted that it is only
the number of users sending text within the same moment that causes
jerkiness, not the total number of users with RTT capability.
3.19. Security for session control and media
Security SHOULD be applied when possible regarding the capabilities
of the participating devices by use of SIP over TLS by default
according to [RFC5630] section 3.1.3 on the session control level and
by default using DTLS-SRTP [RFC5764] on the media level. In
applications where legacy endpoints without security may exist, a
negotiation SHOULD be performed to decide if encryption on the media
level will be applied. If no other security solution is mandated for
the application, then OSRTP [RFC8643] is a suitable method to be
applied to negotiate SRTP media security with DTLS. Most SDP
examples below are for simplicity expressed without the security
additions. The principles (but not all details) for applying DTLS-
SRTP [RFC5764] security are shown in a couple of the following
examples.
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This document contains two mixing procedures which imply different
security levels. The mixing for conference-unaware endpoints has
lower security level than the mixing method for conference-aware
endpoints, because there may be an opportunity for a malicious mixer
or a middleman to masquerade the source labels accompanying the text
streams in text format. This is especially true if support of un-
encrypted SIP and media is supported because of lack of such support
in the target endpoints. However, the mixing for conference-aware
endpoints as specified here also requires that the mixer can be
trusted. Further general security considerations are covered in
Section 11.
End-to-end encryption would require further work and could be based
on WebRTC as specified in Section 1.2.
3.20. SDP offer/answer examples
This section shows some examples of SDP for session negotiation of
the real-time text media in SIP sessions. Audio is usually provided
in the same session, and sometimes also video. The examples only
show the part of importance for the real-time text media. The
examples relate to the single RTP stream mixing for multiparty-aware
endpoints and for multiparty-unaware endpoints.
Note: Multiparty RTT MAY also be provided through other methods,
e.g., by a Selective Forwarding Middlebox (SFM). In that case, the
SDP of the offer will include something specific for that method,
e.g., an SDP attribute or another media format. An answer selecting
the use of that method would accept it by a corresponding
acknowledgement included in the SDP. The offer may contain also the
"rtt-mixer" SDP media attribute for the main RTT media when the
offeror has capability for both multiparty methods, while an answer,
selecting to use SFM will not include the "rtt-mixer" SDP media
attribute.
Offer example for "text/red" format and multiparty support:
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
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Answer example from a multiparty-aware device
m=text 14000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Offer example for "text/red" format including multiparty
and security:
a=fingerprint: (fingerprint1)
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
The "fingerprint" is sufficient to offer DTLS-SRTP, with the media
line still indicating RTP/AVP.
Note: For brevity, the entire value of the SDP fingerprint attribute
is not shown in this and the following example.
Answer example from a multiparty-aware device with security
a=fingerprint: (fingerprint2)
m=text 16000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
With the "fingerprint" the device acknowledges use of SRTP/DTLS.
Answer example from a multiparty-unaware device that also
does not support security:
m=text 12000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
3.21. Packet sequence example from interleaved transmission
This example shows a symbolic flow of packets from a mixer including
loss and recovery. The sequence includes interleaved transmission of
text from two RTT sources A and B. P indicates primary data. R1 is
first redundant generation data and R2 is the second redundant
generation data. A1, B1, A2 etc. are text chunks (T140blocks)
received from the respective sources and sent on to the receiver by
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the mixer. X indicates a dropped packet between the mixer and a
receiver. The session is assumed to use original and two redundant
generations of RTT.
|-----------------------|
|Seq no 101, Time=20400 |
|CC=1 |
|CSRC list A |
|R2: A1, Offset=600 |
|R1: A2, Offset=300 |
|P: A3 |
|-----------------------|
Assuming that earlier packets (with text A1 and A2) were received in
sequence, text A3 is received from packet 101 and assigned to
reception buffer A. The mixer is now assumed to have received
initial text from source B 100 ms after packet 101 and will send that
text. Transmission of A2 and A3 as redundancy is planned for 330 ms
after packet 101 if no new text from A is ready to be sent before
that.
|-----------------------|
|Seq no 102, Time=20500 |
|CC=1 |
|CSRC list B |
|R2 Empty, Offset=600 |
|R1: Empty, Offset=300 |
|P: B1 |
|-----------------------|
Packet 102 is received.
B1 is retrieved from this packet. Redundant transmission of
B1 is planned 330 ms after packet 102.
X------------------------|
X Seq no 103, Timer=20730|
X CC=1 |
X CSRC list A |
X R2: A2, Offset=630 |
X R1: A3, Offset=330 |
X P: Empty |
X------------------------|
Packet 103 is assumed to be lost due to network problems.
It contains redundancy for A. Sending A3 as second level
redundancy is planned for 330 ms after packet 103.
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X------------------------|
X Seq no 104, Timer=20830|
X CC=1 |
X CSRC list B |
X R2: Empty, Offset=600 |
X R1: B1, Offset=300 |
X P: B2 |
X------------------------|
Packet 104 contains text from B, including new B2 and
redundant B1. It is assumed dropped due to network
problems.
The mixer has A3 redundancy to send but no new text
appears from A and therefore the redundancy is sent
330 ms after the previous packet with text from A.
|------------------------|
| Seq no 105, Timer=21060|
| CC=1 |
| CSRC list A |
| R2: A3, Offset=660 |
| R1: Empty, Offset=330 |
| P: Empty |
|------------------------|
Packet 105 is received.
A gap for lost packets 103 and 104 is detected.
Assume that no other loss was detected during the last second.
Then it can be concluded that nothing was totally lost.
R2 is checked. Its original time was 21060-660=20400.
A packet with text from A was received with that
timestamp, so nothing needs to be recovered.
B1 and B2 still need to be transmitted as redundancy.
This is planned 330 ms after packet 105. That
would be at 21160.
|-----------------------|
|Seq no 106, Timer=21160|
|CC=1 |
|CSRC list B |
| R2: B1, Offset=660 |
| R1: B2, Offset=330 |
| P: Empty |
|-----------------------|
Packet 106 is received.
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The second level redundancy in packet 106 is B1 and has timestamp
offset 660 ms. The timestamp of packet 106 minus 660 is 20500 which
is the timestamp of packet 102 THAT was received. So B1 does not
need to be retrieved. The first level redundancy in packet 106 has
offset 330. The timestamp of packet 106 minus 330 is 20830. That is
later than the latest received packet with source B. Therefore B2 is
retrieved and assigned to the input buffer for source B. No primary
is available in packet 106.
After this sequence, A3 and B1 and B2 have been received. In this
case no text was lost.
3.22. Maximum character rate "CPS"
The default maximum rate of reception of "text/t140" real-time text
is in [RFC4103] specified to be 30 characters per second. The value
MAY be modified in the "CPS" parameter of the FMTP attribute in the
media section for the "text/t140" media. A mixer combining real-time
text from a number of sources may occasionally have a higher combined
flow of text coming from the sources. Endpoints SHOULD therefore
specify a suitable higher value for the "CPS" parameter,
corresponding to its real reception capability. A value for "CPS" of
90 SHALL be the default for the "text/t140" stream in the "text/red"
format when multiparty real-time text is negotiated. See [RFC4103]
for the format and use of the "CPS" parameter. The same rules apply
for the multiparty case except for the default value.
4. Presentation level considerations
"Protocol for multimedia application text conversation" [T140]
provides the presentation level requirements for the [RFC4103]
transport. Functions for erasure and other formatting functions are
specified in [T140] which has the following general statement for the
presentation:
"The display of text from the members of the conversation should be
arranged so that the text from each participant is clearly readable,
and its source and the relative timing of entered text is visualized
in the display. Mechanisms for looking back in the contents from the
current session should be provided. The text should be displayed as
soon as it is received."
Strict application of [T140] is of essence for the interoperability
of real-time text implementations and to fulfill the intention that
the session participants have the same information conveyed in the
text contents of the conversation without necessarily having the
exact same layout of the conversation.
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[T140] specifies a set of presentation control codes to include in
the stream. Some of them are optional. Implementations MUST be able
to ignore optional control codes that they do not support.
There is no strict "message" concept in real-time text. The Unicode
Line Separator character SHALL be used as a separator allowing a part
of received text to be grouped in presentation. The characters
"CRLF" may be used by other implementations as a replacement for Line
Separator. The "CRLF" combination SHALL be erased by just one
erasing action, the same as the Line Separator. Presentation
functions are allowed to group text for presentation in smaller
groups than the line separators imply and present such groups with
source indication together with text groups from other sources (see
the following presentation examples). Erasure has no specific limit
by any delimiter in the text stream.
4.1. Presentation by multiparty-aware endpoints
A multiparty-aware receiving party, presenting real-time text MUST
separate text from different sources and present them in separate
presentation fields. The receiving party MAY separate presentation
of parts of text from a source in readable groups based on other
criteria than line separator and merge these groups in the
presentation area when it benefits the user to most easily find and
read text from the different participants. The criteria MAY e.g., be
a received comma, full stop, or other phrase delimiters, or a long
pause.
When text is received from multiple original sources, the
presentation SHALL provide a view where text is added in multiple
presentation fields.
If the presentation presents text from different sources in one
common area, the presenting endpoint SHOULD insert text from the
local user ended at suitable points merged with received text to
indicate the relative timing for when the text groups were completed.
In this presentation mode, the receiving endpoint SHALL present the
source of the different groups of text. This presentation style is
called the "chat" style here and provides a possibility to follow
text arriving from multiple parties and the approximate relative time
that text is received related to text from the local user.
A view of a three-party RTT call in chat style is shown in this
example .
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_________________________________________________
| |^|
|[Alice] Hi, Alice here. |-|
| | |
|[Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris. | |
| I thought you should be here. | |
| | |
|[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|[Bob] And I on Wednesday evening. | |
| | |
|[Alice] Can we meet on Thursday evening? | |
| | |
|[Eve] Yes, definitely. How about 7pm. | |
| at the entrance of the restaurant | |
| Le Lion Blanc? | |
|[Eve] we can have dinner and then take a walk |-|
|______________________________________________|v|
| <Eve-typing> But I need to be back to |^|
| the hotel by 11 because I need |-|
| | |
| <Bob-typing> I wou |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 3: Example of a three-party RTT call presented in chat style
seen at participant 'Alice's endpoint.
Other presentation styles than the chat style MAY be arranged.
This figure shows how a coordinated column view MAY be presented.
_____________________________________________________________________
| Bob | Eve | Alice |
|____________________|______________________|_______________________|
| | |I will arrive by TGV. |
|My flight is to Orly| |Convenient to the main |
| |Hi all, can we plan |station. |
| |for the seminar? | |
|Eve, will you do | | |
|your presentation on| | |
|Friday? |Yes, Friday at 10. | |
|Fine, wo | |We need to meet befo |
|___________________________________________________________________|
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Figure 4: An example of a coordinated column-view of a three-party
session with entries ordered vertically in approximate time-order.
4.2. multiparty mixing for multiparty-unaware endpoints
When the mixer has indicated RTT multiparty capability in an SDP
negotiation, but the multiparty capability negotiation fails with an
endpoint, then the agreed "text/red" or "text/t140" format SHALL be
used and the mixer SHOULD compose a best-effort presentation of
multiparty real-time text in one stream intended to be presented by
an endpoint with no multiparty awareness, when that is desired in the
actual implementation. The following specifies a procedure which MAY
be applied in that situation.
This presentation format has functional limitations and SHOULD be
used only to enable participation in multiparty calls by legacy
deployed endpoints implementing only RFC 4103 without any multiparty
extensions specified in this document.
The principles and procedures below do not specify any new protocol
elements. They are instead composed from the information in [T140]
and an ambition to provide a best-effort presentation on an endpoint
which has functions only for two-party calls.
The mixer mixing for multiparty-unaware endpoints SHALL compose a
simulated, limited multiparty RTT view suitable for presentation in
one presentation area. The mixer SHALL group text in suitable groups
and prepare for presentation of them by inserting a new line between
them if the transmitted text did not already end with a new line. A
presentable label SHALL be composed and sent for the source initially
in the session and after each source switch. With this procedure the
time for switching from transmission of text from one source to
transmission of text from another source depends on the actions of
the users. In order to expedite source switching, a user can, for
example, end its turn with a new line.
4.2.1. Actions by the mixer at reception from the call participants
When text is received by the mixer from the different participants,
the mixer SHALL recover text from redundancy if any packets are lost.
The mark for lost text [T140ad1] SHALL be inserted in the stream if
unrecoverable loss appears. Any Unicode "BOM" characters, possibly
used for keep-alive, SHALL be deleted. The time of creation of text
(retrieved from the RTP timestamp) SHALL be stored together with the
received text from each source in queues for transmission to the
recipients in order to be able to evaluate text loss.
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4.2.2. Actions by the mixer for transmission to the recipients
The following procedure SHALL be applied for each multiparty-unaware
recipient of multiparty text from the mixer.
The text for transmission SHALL be formatted by the mixer for each
receiving user for presentation in one single presentation area.
Text received from a participant SHOULD NOT be included in
transmission to that participant because it is usually presented
locally at transmission time. When there is text available for
transmission from the mixer to a receiving party from more than one
participant, the mixer SHALL switch between transmission of text from
the different sources at suitable points in the transmitted stream.
When switching source, the mixer SHALL insert a line separator if the
already transmitted text did not end with a new line (line separator
or CRLF). A label SHALL be composed from information in the CNAME
and NAME fields in RTCP reports from the participant to have its text
transmitted, or from other session information for that user. The
label SHALL be delimited by suitable characters (e.g., '[ ]') and
transmitted. The CSRC SHALL indicate the selected source. Then text
from that selected participant SHALL be transmitted until a new
suitable point for switching source is reached.
Information available to the mixer for composing the label may
contain sensitive personal information that SHOULD not be revealed in
sessions not securely authenticated and protected. Integrity
considerations regarding how much personal information is included in
the label SHOULD therefore be taken when composing the label.
Seeking a suitable point for switching source SHALL be done when
there is older text waiting for transmission from any party than the
age of the last transmitted text. Suitable points for switching are:
* A completed phrase ended by comma
* A completed sentence
* A new line (line separator or CRLF)
* A long pause (e.g., > 10 seconds) in received text from the
currently transmitted source
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* If text from one participant has been transmitted with text from
other sources waiting for transmission for a long time (e.g., > 1
minute) and none of the other suitable points for switching has
occurred, a source switch MAY be forced by the mixer at the next
word delimiter, and also even if a word delimiter does not occur
within a time (e.g., 15 seconds) after the scan for a word
delimiter started.
When switching source, the source which has the oldest text in queue
SHALL be selected to be transmitted. A character display count SHALL
be maintained for the currently transmitted source, starting at zero
after the label is transmitted for the currently transmitted source.
The status SHALL be maintained for the latest control code for Select
Graphic Rendition (SGR) from each source. If there is an SGR code
stored as the status for the current source before the source switch
is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B
0000 006D] after the new line and before the new label during a
source switch. See SGR below for an explanation. This transmission
does not influence the display count.
If there is an SGR code stored for the new source after the source
switch, that SGR code SHALL be transmitted to the recipient before
the label. This transmission does not influence the display count.
4.2.3. Actions on transmission of text
Text from a source sent to the recipient SHALL increase the display
count by one per transmitted character.
4.2.4. Actions on transmission of control codes
The following control codes specified by T.140 require specific
actions. They SHALL cause specific considerations in the mixer.
Note that the codes presented here are expressed in UCS-16, while
transmission is made in the UTF-8 encoding of these codes.
BEL 0007 Bell Alert in session. Provides for alerting during an
active session. The display count SHALL NOT be altered.
NEW LINE 2028 Line separator. Check and perform a source switch if
appropriate. Increase the display count by 1.
CR LF 000D 000A A supported but not preferred way of requesting a
new line. Check and perform a source switch if appropriate.
Increase the display count by 1.
INT ESC 0061 Interrupt (used to initiate the mode negotiation
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procedure). The display count SHALL NOT be altered.
SGR 009B Ps 006D Select graphic rendition. Ps is the rendition
parameters specified in ISO 6429. The display count SHALL NOT be
altered. The SGR code SHOULD be stored for the current source.
SOS 0098 Start of string, used as a general protocol element
introducer, followed by a maximum 256-byte string and the ST. The
display count SHALL NOT be altered.
ST 009C String terminator, end of SOS string. The display count
SHALL NOT be altered.
ESC 001B Escape - used in control strings. The display count SHALL
NOT be altered for the complete escape code.
Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used
for synchronization and keep-alive. It SHALL be deleted from
incoming streams. It SHALL also be sent first after session
establishment to the recipient. The display count SHALL NOT be
altered.
Missing text mark (U+FFFD) "Replacement character", represented as a
question mark in a rhombus, or if that is not feasible, replaced
by an apostrophe '. It marks the place in the stream of possible
text loss. This mark SHALL be inserted by the reception procedure
in case of unrecoverable loss of packets. The display count SHALL
be increased by one when sent as for any other character.
SGR If a control code for selecting graphic rendition (SGR) other
than reset of the graphic rendition (SGR 0) is sent to a
recipient, that control code SHALL also be stored as the status
for the source in the storage for SGR status. If a reset graphic
rendition (SGR 0) originating from a source is sent, then the SGR
status storage for that source SHALL be cleared. The display
count SHALL NOT be increased.
BS (U+0008) Back Space, intended to erase the last entered character
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by a source. Erasure by backspace cannot always be performed as
the erasing party intended. If an erasing action erases all text
up to the end of the leading label after a source switch, then the
mixer MUST NOT transmit more backspaces. Instead it is
RECOMMENDED that a letter "X" is inserted in the text stream for
each backspace as an indication of the intent to erase more. A
new line is usually coded by a Line Separator, but the character
combination "CRLF" MAY be used instead. Erasure of a new line is
in both cases done by just one erasing action (Backspace). If the
display count has a positive value it SHALL be decreased by one
when the BS is sent. If the display count is at zero, it SHALL
NOT be altered.
4.2.5. Packet transmission
A mixer transmitting to a multiparty-unaware terminal SHALL send
primary data only from one source per packet. The SSRC SHALL be the
SSRC of the mixer. The CSRC list SHALL contain one member and be the
SSRC of the source of the primary data.
4.2.6. Functional limitations
When a multiparty-unaware endpoint presents a conversation in one
display area in a chat style, it inserts source indications for
remote text and local user text as they are merged in completed text
groups. When an endpoint using this layout receives and presents
text mixed for multiparty-unaware endpoints, there will be two levels
of source indicators for the received text; one generated by the
mixer and inserted in a label after each source switch, and another
generated by the receiving endpoint and inserted after each switch
between local and remote source in the presentation area. This will
waste display space and look inconsistent to the reader.
New text can be presented only from one source at a time. Switch of
source to be presented takes place at suitable places in the text,
such as end of phrase, end of sentence, line separator and
inactivity. Therefore the time to switch to present waiting text
from other sources may become long and will vary and depend on the
actions of the currently presented source.
Erasure can only be done up to the latest source switch. If a user
tries to erase more text, the erasing actions will be presented as
letter X after the label.
Text loss because of network errors may hit the label between entries
from different parties, causing risk for misunderstanding from which
source a piece of text is.
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These facts make it strongly RECOMMENDED to implement multiparty
awareness in RTT endpoints. The use of the mixing method for
multiparty-unaware endpoints should be left for use with endpoints
which are impossible to upgrade to become multiparty-aware.
4.2.7. Example views of presentation on multiparty-unaware endpoints
The following pictures are examples of the view on a participant's
display for the multiparty-unaware case.
_________________________________________________
| Conference | Alice |
|________________________|_________________________|
| |I will arrive by TGV. |
|[Bob]:My flight is to |Convenient to the main |
|Orly. |station. |
|[Eve]:Hi all, can we | |
|plan for the seminar. | |
| | |
|[Bob]:Eve, will you do | |
|your presentation on | |
|Friday? | |
|[Eve]:Yes, Friday at 10.| |
|[Bob]: Fine, wo |We need to meet befo |
|________________________|_________________________|
Figure 5: Alice who has a conference-unaware client is receiving the
multiparty real-time text in a single-stream.
This figure shows how a coordinated column view MAY be presented on
Alice's device in a view with two-columns. The mixer inserts labels
to show how the sources alternate in the column with received text.
The mixer alternates between the sources at suitable points in the
text exchange so that text entries from each party can be
conveniently read.
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_________________________________________________
| |^|
|(Alice) Hi, Alice here. |-|
| | |
|(mix)[Bob)] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris | |
| I thought you should be here. | |
| | |
|(Alice) I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|(mix)[Bob] And I on Wednesday evening. | |
| | |
|[Eve] we can have dinner and then walk | |
| | |
|[Eve] But I need to be back to | |
| the hotel by 11 because I need | |
| |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 6: An example of a view of the multiparty-unaware presentation
in chat style. Alice is the local user.
In this view, there is a tradition in receiving applications to
include a label showing the source of the text, here shown with
parenthesis "()". The mixer also inserts source labels for the
multiparty call participants, here shown with brackets "[]".
5. Relation to Conference Control
5.1. Use with SIP centralized conferencing framework
The SIP conferencing framework, mainly specified in [RFC4353],
[RFC4579] and [RFC4575] is suitable for coordinating sessions
including multiparty RTT. The RTT stream between the mixer and a
participant is one and the same during the conference. Participants
get announced by notifications when participants are joining or
leaving, and further user information may be provided. The SSRC of
the text to expect from joined users MAY be included in a
notification. The notifications MAY be used both for security
purposes and for translation to a label for presentation to other
users.
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5.2. Conference control
In managed conferences, control of the real-time text media SHOULD be
provided in the same way as other for media, e.g., for muting and
unmuting by the direction attributes in SDP [RFC8866].
Note that floor control functions may be of value for RTT users as
well as for users of other media in a conference.
6. Gateway Considerations
6.1. Gateway considerations with Textphones
multiparty RTT sessions may involve gateways of different kinds.
Gateways involved in setting up sessions SHALL correctly reflect the
multiparty capability or unawareness of the combination of the
gateway and the remote endpoint beyond the gateway.
One case that may occur is a gateway to PSTN for communication with
textphones (e.g., TTYs). Textphones are limited devices with no
multiparty awareness, and it SHOULD therefore be suitable for the
gateway to not indicate multiparty awareness for that case. Another
solution is that the gateway indicates multiparty capability towards
the mixer, and includes the multiparty mixer function for multiparty-
unaware endpoints itself. This solution makes it possible to adapt
to the functional limitations of the textphone.
More information on gateways to textphones is found in [RFC5194]
6.2. Gateway considerations with WebRTC
Gateway operation to real-time text in WebRTC may also be required.
In WebRTC, RTT is specified in [RFC8865].
A multiparty bridge may have functionality for communicating by RTT
both in RTP streams with RTT and WebRTC T.140 data channels. Other
configurations may consist of a multiparty bridge with either
technology for RTT transport and a separate gateway for conversion of
the text communication streams between RTP and T.140 data channel.
In WebRTC, it is assumed that for a multiparty session, one T.140
data channel is established for each source from a gateway or bridge
to each participant. Each participant also has a data channel with a
two-way connection with the gateway or bridge.
The T.140 data channel used both ways is for text from the WebRTC
user and from the bridge or gateway itself to the WebRTC user. The
label parameter of this T.140 data channel is used as the NAME field
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in RTCP to participants on the RTP side. The other T.140 data
channels are only for text from other participants to the WebRTC
user.
When a new participant has entered the session with RTP transport of
RTT, a new T.140 channel SHOULD be established to WebRTC users with
the label parameter composed from the NAME field in RTCP on the RTP
side.
When a new participant has entered the multiparty session with RTT
transport in a WebRTC T.140 data channel, the new participant SHOULD
be announced by a notification to RTP users. The label parameter
from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP
side, or other available session information.
When a participant on the RTP side is disconnected from the
multiparty session, the corresponding T.140 data channel(s) SHOULD be
closed.
When a WebRTC user of T.140 data channels disconnects from the mixer,
the corresponding RTP streams or sources in an RTP-mixed stream
SHOULD be closed.
T.140 data channels MAY be opened and closed by negotiation or
renegotiation of the session or by any other valid means as specified
in section 1 of [RFC8865].
7. Updates to RFC 4103
This document updates [RFC4103] by introducing an SDP media attribute
"rtt-mixer" for negotiation of multiparty-mixing capability with the
[RFC4103] format, and by specifying the rules for packets when
multiparty capability is negotiated and in use.
8. Congestion considerations
The congestion considerations and recommended actions from [RFC4103]
are also valid in multiparty situations.
The first action in case of congestion SHALL be to temporarily
increase the transmission interval up to two seconds.
If the very unlikely situation appears that many participants in a
conference send text simultaneously for a long period, a delay may
build up for presentation of text at the receivers if the limitation
in characters per second ("CPS") to be transmitted to the
participants is exceeded. More delay than 7 seconds can cause
confusion in the session. It is therefore RECOMMENDED that an RTP-
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mixer-based mixer discards such text causing excessive delays and
inserts a general indication of possible text loss [T140ad1] in the
session. If the main text contributor is indicated in any way, the
mixer MAY avoid deleting text from that participant. It should
however be noted that human creation of text normally contains
pauses, when the transmission can catch up, so that the transmission
overload situations are expected to be very rare.
9. Acknowledgements
James Hamlin for format and performance aspects.
10. IANA Considerations
10.1. Registration of the "rtt-mixer" SDP media attribute
[RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
RFC number of this document.]
IANA is asked to register the new SDP attribute "rtt-mixer".
Contact name: IESG
Contact email: iesg@ietf.org
Attribute name: rtt-mixer
Attribute semantics: See RFCXXXX Section 2.3
Attribute value: none
Usage level: media
Purpose: Indicate support by mixer and endpoint of multiparty mixing
for real-time text transmission, using a common RTP-stream for
transmission of text from a number of sources mixed with one
source at a time and the source indicated in a single CSRC-list
member.
Charset Dependent: no
O/A procedure: See RFCXXXX Section 2.3
Mux Category: normal
Reference: RFCXXXX
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11. Security Considerations
The RTP-mixer model requires the mixer to be allowed to decrypt,
pack, and encrypt secured text from the conference participants.
Therefore the mixer needs to be trusted to achieve security in
confidentiality and integrity. This situation is similar to the
situation for handling audio and video media in centralized mixers.
The requirement to transfer information about the user in RTCP
reports in SDES, CNAME, and NAME fields, and in conference
notifications, for creation of labels may have privacy concerns as
already stated in RFC 3550 [RFC3550], and may be restricted for
privacy reasons. The receiving user will then get a more symbolic
label for the source.
Participants with malicious intentions may appear and e.g., disturb
the multiparty session by emitting a continuous flow of text. They
may also send text that appears to originate from other participants.
Counteractions should be to require secure signaling, media and
authentication, and to provide higher-layer conference functions
e.g., for blocking, muting, and expelling participants.
Further security considerations specific for this application are
specified in Section 3.19.
12. Change history
[RFC Editor: Please remove this section prior to publication.]
12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17
Actions on Gen-ART review comments.
Actions on SecDir review comments.
12.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16
Improvements in the offer/answer considerations section by adding
subsections for each phase in the negotiation as requested by IANA
expert review.
12.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15
Actions on review comments from Jurgen Schonwalder:
A bit more about congestion situations and that they are expected to
be very rare.
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Explanation of differences in security between the conference-aware
and the conference-unaware case added in security section.
Presentation examples with suource labels made less confusing, and
explained.
Reference to T.140 inserted at first mentioning of T.140.
Reference to RFC 8825 inserted to explain WebRTC
Nit in wording in terminology section adjusted.
12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14
Changes from comments by Murray Kucherawy during AD review.
Many SHOULD in section 4.2 on multiparty-unaware mixing changed to
SHALL, and the whole section instead specified to be optional
depending on the application.
Some SHOULD in section 3 either explained or changed to SHALL.
In order to have explainable conditions behind SHOULDs, the
transmission interval in 3.4 is changed to as soon as text is
available as a main principle. The call participants send with 300
ms interval so that will create realistic load conditions anyway.
12.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13
Changed year to 2021.
Changed reference to draft on RTT in WebRTC to recently published RFC
8865.
Changed label brackets in example from "[]" to "()" to avoid nits
comment.
Changed reference "RFC 4566" to recently published "RFC 8866"
12.6. 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.
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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.
12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11
Timestamps and timestamp offsets added to the packet examples in
section 3.23, and the description corrected.
A number of minor corrections added in sections 3.10 - 3.23.
12.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10
The packet composition was modified for interleaving packets from
different sources.
The packet reception was modified for the new interleaving method.
The packet sequence examples was adjusted for the new interleaving
method.
Modifications according to responses to Brian Rosen of 2020-11-03
12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09
Changed name on the SDP media attribute to "rtt-mixer"
Restructure of section 2 for balance between aware and unaware cases.
Moved conference control to own section.
Improved clarification of recovery and loss in the packet sequence
example.
A number of editorial corrections and improvements.
12.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08
Deleted the method requiring a new packet format "text/rex" because
of the longer standardization and implementation period it needs.
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Focus on use of RFC 4103 text/red format with shorter transmission
interval, and source indicated in CSRC.
12.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07
Added a method based on the "text/red" format and single source per
packet, negotiated by the "rtt-mixer" SDP attribute.
Added reasoning and recommendation about indication of loss.
The highest number of sources in one packet is 15, not 16. Changed.
Added in information on update to RFC 4103 that RFC 4103 explicitly
allows addition of FEC method. The redundancy is a kind of forward
error correction..
12.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06
Improved definitions list format.
The format of the media subtype parameters is made to match the
requirements.
The mapping of media subtype parameters to SDP is included.
The "CPS" parameter belongs to the t140 subtype and does not need to
be registered here.
12.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05
nomenclature and editorial improvements
"this document" used consistently to refer to this document.
12.14. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04
'Redundancy header' renamed to 'data header'.
More clarifications added.
Language and figure number corrections.
12.15. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03
Mention possible need to mute and raise hands as for other media.
---done ----
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Make sure that use in two-party calls is also possible and explained.
- may need more wording -
Clarify the RTT is often used together with other media. --done--
Tell that text mixing is N-1. A users own text is not received in
the mix. -done-
In 3. correct the interval to: A "text/rex" transmitter SHOULD send
packets distributed in time as long as there is something (new or
redundant T140blocks) to transmit. The maximum transmission interval
SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a
receiver as soon as new text to that receiver is available, as long
as the time after the latest sent packet to the same receiver is more
than 150 ms, and also the maximum character rate to the receiver is
not exceeded. The intention is to keep the latency low while keeping
a good protection against text loss in bursty packet loss conditions.
-done-
In 1.3 say that the format is used both ways. -done-
In 13.1 change presentation area to presentation field so that reader
does not think it shall be totally separated. -done-
In Performance and intro, tell the performance in number of
simultaneous sending users and introduced delay 16, 150 vs
requirements 5 vs 500. -done --
Clarify redundancy level per connection. -done-
Timestamp also for the last data header. To make it possible for all
text to have time offset as for transmission from the source. Make
that header equal to the others. -done-
Mixer always use the CSRC list, even for its own BOM. -done-
Combine all talk about transmission interval (300 ms vs when text has
arrived) in section 3 in one paragraph or close to each other. -done-
Documents the goal of good performance with low delay for 5
simultaneous typers in the introduction. -done-
Describe better that only primary text shall be sent on to receivers.
Redundancy and loss must be resolved by the mixer. -done-
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12.16. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02
SDP and better description and visibility of security by OSRTP RFC
8634 needed.
The description of gatewaying to WebRTC extended.
The description of the data header in the packet is improved.
12.17. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01
2,5,6 More efficient format "text/rex" introduced and attribute
a=rtt-mix deleted.
3. Brief about use of OSRTP for security included- More needed.
4. Brief motivation for the solution and why not rtp-translator is
used added to intro.
7. More limitations for the multiparty-unaware mixing method
inserted.
8. Updates to RFC 4102 and 4103 more clearly expressed.
9. Gateway to WebRTC started. More needed.
12.18. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03
to draft-ietf-avtcore-multi-party-rtt-mix-00
Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00
Replaced CDATA in IANA registration table with better coding.
Converted to xml2rfc version 3.
12.19. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02
to -03
Changed company and e-mail of the author.
Changed title to "RTP-mixer formatting of multi-party Real-time text"
to better match contents.
Check and modification where needed of use of RFC 2119 words SHALL
etc.
More about the CC value in sections on transmitters and receivers so
that 1-to-1 sessions do not use the mixer format.
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Enhanced section on presentation for multiparty-unaware endpoints
A paragraph recommending CPS=150 inserted in the performance section.
12.20. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01
to -02
In Abstract and 1. Introduction: Introduced wording about regulatory
requirements.
In section 5: The transmission interval is decreased to 100 ms when
there is text from more than one source to transmit.
In section 11 about SDP negotiation, a SHOULD-requirement is
introduced that the mixer should make a mix for multiparty-unaware
endpoints if the negotiation is not successful. And a reference to a
later chapter about it.
The presentation considerations chapter 14 is extended with more
information about presentation on multiparty-aware endpoints, and a
new section on the multiparty-unaware mixing with low functionality
but SHOULD a be implemented in mixers. Presentation examples are
added.
A short chapter 15 on gateway considerations is introduced.
Clarification about the text/t140 format included in chapter 10.
This sentence added to the chapter 10 about use without redundancy.
"The text/red format SHOULD be used unless some other protection
against packet loss is utilized, for example a reliable network or
transport."
Note about deviation from RFC 2198 added in chapter 4.
In chapter 9. "Use with SIP centralized conferencing framework" the
following note is inserted: Note: The CSRC-list in an RTP packet only
includes participants who's text is included in one or more text
blocks. It is not the same as the list of participants in a
conference. With audio and video media, the CSRC-list would often
contain all participants who are not muted whereas text participants
that don't type are completely silent and so don't show up in RTP
packet CSRC-lists.
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12.21. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00
to -01
Editorial cleanup.
Changed capability indication from fmtp-parameter to SDP attribute
"rtt-mix".
Swapped order of redundancy elements in the example to match reality.
Increased the SDP negotiation section
13. References
13.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type",
RFC 4102, DOI 10.17487/RFC4102, June 2005,
<https://www.rfc-editor.org/info/rfc4102>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630,
DOI 10.17487/RFC5630, October 2009,
<https://www.rfc-editor.org/info/rfc5630>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
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[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011,
<https://www.rfc-editor.org/info/rfc6263>.
[RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
Thomson, "Session Traversal Utilities for NAT (STUN) Usage
for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
October 2015, <https://www.rfc-editor.org/info/rfc7675>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
[RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text
Conversation over WebRTC Data Channels", RFC 8865,
DOI 10.17487/RFC8865, January 2021,
<https://www.rfc-editor.org/info/rfc8865>.
[RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP:
Session Description Protocol", RFC 8866,
DOI 10.17487/RFC8866, January 2021,
<https://www.rfc-editor.org/info/rfc8866>.
[T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for
multimedia application text conversation", February 1998,
<https://www.itu.int/rec/T-REC-T.140-199802-I/en>.
[T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000),
Protocol for multimedia application text conversation",
February 2000,
<https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>.
13.2. Informative References
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353,
DOI 10.17487/RFC4353, February 2006,
<https://www.rfc-editor.org/info/rfc4353>.
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[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <https://www.rfc-editor.org/info/rfc4575>.
[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents",
BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006,
<https://www.rfc-editor.org/info/rfc4579>.
[RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real-
Time Text over IP Using the Session Initiation Protocol
(SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008,
<https://www.rfc-editor.org/info/rfc5194>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T.
Stach, "An Opportunistic Approach for Secure Real-time
Transport Protocol (OSRTP)", RFC 8643,
DOI 10.17487/RFC8643, August 2019,
<https://www.rfc-editor.org/info/rfc8643>.
Author's Address
Gunnar Hellstrom
Gunnar Hellstrom Accessible Communication
SE-13670 Vendelso
Sweden
Email: gunnar.hellstrom@ghaccess.se
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