RTP-mixer formatting of multi-party Real-time text
draft-ietf-avtcore-multi-party-rtt-mix-11
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 | 2020-12-01 (Latest revision 2020-11-22) | ||
| Replaces | draft-hellstrom-avtcore-multi-party-rtt-source | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
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| Additional resources | Mailing list discussion | ||
| Stream | WG state | In WG Last Call | |
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| Document shepherd | (None) | ||
| IESG | IESG state | Became RFC 9071 (Proposed Standard) | |
| Consensus boilerplate | Yes | ||
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draft-ietf-avtcore-multi-party-rtt-mix-11
AVTCore G. Hellstrom
Internet-Draft Gunnar Hellstrom Accessible Communication
Updates: RFC 4103 (if approved) 22 November 2020
Intended status: Standards Track
Expires: 26 May 2021
RTP-mixer formatting of multi-party Real-time text
draft-ietf-avtcore-multi-party-rtt-mix-11
Abstract
Real-time text mixers for multi-party sessions need to identify the
source of each transmitted group of text so that the text can be
presented by endpoints in suitable grouping with other text from the
same source, while new text from other sources is also presented in
readable grouping as received interleaved in real-time.
Regional regulatory requirements specify provision of real-time text
in multi-party calls. RFC 4103 mixer implementations can use
traditional RTP functions for source identification, but the mixer
source switching performance is limited when using the default
transmission characteristics with redundancy.
Enhancements for RFC 4103 real-time text mixing is provided in this
document, suitable for a centralized conference model that enables
source identification and source switching. The intended use is for
real-time text mixers and multi-party-aware participant endpoints.
The specified mechanism build on the standard use of the CSRC list in
the RTP packet for source identification. The method makes use of
the same "text/t140" and "text/red" formats as for two-party
sessions.
A capability exchange is specified so that it can be verified that a
participant can handle the multi-party coded real-time text stream.
The capability is indicated by use of a media attribute "rtt-mixer".
The document updates RFC 4103[RFC4103]
A specifications of how a mixer can format text for the case when the
endpoint is not multi-party aware is also provided.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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Internet-Drafts are working documents of the Internet Engineering
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working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on 26 May 2021.
Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/
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Please review these documents carefully, as they describe your rights
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provided without warranty as described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Selected solution and considered alternative . . . . . . 5
1.2. Nomenclature . . . . . . . . . . . . . . . . . . . . . . 7
1.3. Intended application . . . . . . . . . . . . . . . . . . 8
2. Overview over the two specified solutions . . . . . . . . . . 9
2.1. Negotiated use of the RFC 4103 format for multi-party
transmission in a single RTP stream . . . . . . . . . . . 9
2.2. Mixing for multi-party unaware endpoints . . . . . . . . 9
3. Details for the multi-party aware mixing case . . . . . . . . 9
3.1. Offer/answer considerations . . . . . . . . . . . . . . . 10
3.2. Actions depending on capability negotiation result . . . 10
3.3. Use of fields in the RTP packets . . . . . . . . . . . . 10
3.4. Initial transmission of a BOM character . . . . . . . . . 11
3.5. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 11
3.6. Transmission interval . . . . . . . . . . . . . . . . . . 11
3.7. Only one source per packet . . . . . . . . . . . . . . . 11
3.8. Do not send received text to the originating source . . . 12
3.9. Clean incoming text . . . . . . . . . . . . . . . . . . . 12
3.10. Redundant transmission principles . . . . . . . . . . . . 12
3.11. Interleaving text from different sources . . . . . . . . 12
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3.12. Text placement in packets . . . . . . . . . . . . . . . . 12
3.13. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 13
3.14. Creation of the redundancy . . . . . . . . . . . . . . . 13
3.15. Timer offset fields . . . . . . . . . . . . . . . . . . . 14
3.16. Other RTP header fields . . . . . . . . . . . . . . . . . 14
3.17. Pause in transmission . . . . . . . . . . . . . . . . . . 14
3.18. RTCP considerations . . . . . . . . . . . . . . . . . . . 15
3.19. Reception of multi-party contents . . . . . . . . . . . . 15
3.20. Performance considerations . . . . . . . . . . . . . . . 17
3.21. Security for session control and media . . . . . . . . . 18
3.22. SDP offer/answer examples . . . . . . . . . . . . . . . . 18
3.23. Packet sequence example from interleaved transmission . . 19
3.24. Maximum character rate "CPS" . . . . . . . . . . . . . . 22
4. Presentation level considerations . . . . . . . . . . . . . . 22
4.1. Presentation by multi-party aware endpoints . . . . . . . 23
4.2. Multi-party mixing for multi-party unaware endpoints . . 25
5. Relation to Conference Control . . . . . . . . . . . . . . . 30
5.1. Use with SIP centralized conferencing framework . . . . . 31
5.2. Conference control . . . . . . . . . . . . . . . . . . . 31
6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 31
6.1. Gateway considerations with Textphones (e.g. TTYs). . . 31
6.2. Gateway considerations with WebRTC. . . . . . . . . . . . 32
7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 32
8. Congestion considerations . . . . . . . . . . . . . . . . . . 32
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
10.1. Registration of the "rtt-mixer" sdp media attribute . . 33
11. Security Considerations . . . . . . . . . . . . . . . . . . . 34
12. Change history . . . . . . . . . . . . . . . . . . . . . . . 34
12.1. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 34
12.2. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 34
12.3. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 34
12.4. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 35
12.5. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 35
12.6. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 35
12.7. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 35
12.8. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 36
12.9. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 36
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12.10. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 37
12.11. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 37
12.12. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-03 to
draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 37
12.13. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-02 to
-03 . . . . . . . . . . . . . . . . . . . . . . . . . . 38
12.14. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-01 to
-02 . . . . . . . . . . . . . . . . . . . . . . . . . . 38
12.15. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-00 to
-01 . . . . . . . . . . . . . . . . . . . . . . . . . . 39
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 39
13.1. Normative References . . . . . . . . . . . . . . . . . . 39
13.2. Informative References . . . . . . . . . . . . . . . . . 40
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 41
1. Introduction
RFC 4103[RFC4103] specifies use of RFC 3550 RTP [RFC3550] for
transmission of real-time text (RTT) and the "text/t140" format. It
also specifies a redundancy format "text/red" for increased
robustness. RFC 4102 [RFC4102] registers the "text/red" format.
Regional regulatory requirements specify provision of real-time text
in multi-party calls.
Real-time text is usually provided together with audio and sometimes
with video in conversational sessions.
A requirement related to multi-party sessions from the presentation
level standard T.140 for real-time text is: "The display of text from
the members of the conversation should be arranged so that the text
from each participant is clearly readable, and its source and the
relative timing of entered text is visualized in the display."
Another requirement is that the mixing procedure must not introduce
delays in the text streams that are experienced disturbing the real-
time experience of the receiving users.
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The redundancy scheme of RFC 4103 [RFC4103] enables efficient
transmission of redundant text in packets together with new text.
However the redundancy header format has no source indicators for the
redundant transmissions. 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 multi-party sessions therefore need to
insert the source of each transmitted group of text from a conference
participant so that the text can be transmitted interleaved with text
groups from different sources in the rate they are created. This
enables the text groups to be presented by endpoints in suitable
grouping with other text from the same source. The presentation can
then be arranged so that text from different sources can be presented
in real-time and easily read while it is possible for a reading user
to also perceive approximately when the text was created in real time
by the different parties. The transmission and mixing is intended to
be done in a general way so that presentation can be arranged in a
layout decided by the endpoint.
There are existing implementations of RFC 4103 without the updates
from this document. These will not be able to receive and present
real-time text mixed for multi-party aware endpoints.
A negotiation mechanism is therefore needed for verification if the
parties are able to handle a multi-party coded stream and agreeing on
using that method.
A fall-back mixing procedure is also needed for cases when the
negotiation result indicates that a receiving endpoint is not capable
of handling the mixed format. This method is called the mixing
procedure for multi-party unaware endpoints. The fall-back method is
naturally not expected to meet all performance requirements placed on
the mixing procedure for multi-party aware endpoints.
The document updates RFC 4103[RFC4103] by introducing an attribute
for indicating capability for the multi-party mixing case and rules
for source indications and source switching.
1.1. Selected solution and considered alternative
A number of alternatives were considered when searching an efficient
and easily implemented multi-party method for real-time text. This
section explains a few of them briefly.
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One RTP stream per source, sent in the same RTP session with
"text/red" format.
From some points of view, use of multiple RTP streams, one for
each source, sent in the same RTP session, called the RTP
translator model in [RFC3550], would be efficient, and use exactly
the same packet format as [RFC4103], the same payload type and a
simple SDP declaration. However, the RTP implementation in both
mixers and endpoints need to support multiple streams in the same
RTP session in order to use this mechanism. For best deployment
opportunity, it should be possible to upgrade existing endpoint
solutions to be multi-party aware with a reasonable effort. There
is currently a lack of support for multi-stream RTP in certain
implementation technologies. This fact made this solution not
selected for inclusion in this document.
The "text/red" format in RFC 4103 with shorter transmission
interval, and indicating source in CSRC.
The "text/red" format with "text/t140" payload in a single RTP
stream can be sent with 100 ms packet intervals instead of the
regular 300 ms. The source is indicated in the CSRC field.
Source switching can then be done every 100 ms while simultaneous
transmission occurs. With five participants sending text
simultaneously, the switching and transmission performance is
good. With more simultaneously sending participants, there will
be a noticable jerkiness in text presentation. The jerkiness will
be more expressed the more participants who send text
simultaneously. With ten sending participants, the jerkiness will
be about one second. Text sent from a source at the end of the
period its text is sent by the mixer will have close to zero extra
delay. Recent text will be presented with no or low delay. The
one second jerkiness will be noticable and slightly unpleasant,
but corresponds in time to what typing humans often cause by
hesitation or changing position while typing. A benefit of this
method is that no new packet format needs to be introduced and
implemented. Since simultaneous typing by more than two parties
is very rare, and in most applications also more than three
parties in a call is rare, this method can be used successfully
with good performance. Recovery of text in case of packet loss is
based on analysis of timestamps of received redundancy versus
earlier received text. Negotiation is based on a new sdp media
attribute "rtt-mixer". This method is selected to be the main one
specified in this document.
A new "text" media subtype with up to 15 sources in each packet.
The mechanism makes use of the RTP mixer model specified in
RFC3550[RFC3550]. Text from up to 15 sources can be included in
each packet. Packets are normally sent every 300 ms. The mean
delay will be 150 ms. The sources are indicated in strict order
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in the CSRC list of the RTP packets. A new redundancy packet
format is specified. This method would result in good
performance, but would require standardisation and implementation
of new releases in the target technologies that would take more
time than desirable to complete. It was therefore not selected to
be included in this document.
The presentation planned by the mixer for multi-party unaware
endpoints.
It is desirable to have a method that does not require any
modifications in existing user devices implementing RFC 4103 for
RTT without explicit support of multi-party sessions. This is
possible by having the mixer insert a new line and a text
formatted source label before each switch of text source in the
stream. Switch of source can only be done in places in the text
where it does not disturb the perception of the contents. Text
from only one source can be presented in real time at a time. The
delay will therefore be varying. The method has also other
limitations, but is included in this document as a fallback
method. In calls where parties take turns properly by ending
their entries with a new line, the limitations will have limited
influence on the user experience. while only two parties send
text, these two will see the text in real time with no delay.
This method is specified as a fallback method in this document.
RTT transport in WebRTC
Transport of real-time text in the WebRTC technology is specified
to use the WebRTC data channel in
[I-D.ietf-mmusic-t140-usage-data-channel]. That spcification
contains a section briefly describing its use in multi-party
sessions. The focus of this document is RTP transport.
Therefore, even if the WebRTC transport provides good multi-party
performance, it is just mentioned in this document in relation to
providing gateways with multi-party capabilities between RTP and
WebRTC technologies.
1.2. Nomenclature
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
The terms SDES, CNAME, NAME, SSRC, CSRC, CSRC list, CC, RTCP, RTP-
mixer, RTP-translator are explained in [RFC3550]
The term "T140block" is defined in RFC 4103 [RFC4103] to contain one
or more T.140 code elements.
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"TTY" stands for a text telephone type used in North America.
"WebRTC" stands for web based communication specified by W3C and
IETF.
"DTLS-SRTP" stands for security specified in RFC 5764 [RFC5764].
"multi-party aware" stands for an endpoint receiving real-time text
from multiple sources through a common conference mixer being able to
present the text in real-time separated by source and presented so
that a user can get an impression of the approximate relative timing
of text from different parties.
"multi-party unaware" stands for an endpoint not itself being able to
separate text from different sources when received through a common
conference mixer.
1.3. Intended application
The method for multi-party real-time text specified in this document
is primarily intended for use in transmission between mixers and
endpoints in centralised mixing configurations. It is also
applicable between mixers. An often mentioned application is for
emergency service calls with real-time text and voice, where a
calltaker want to make an attended handover of a call to another
agent, and stay observing the session. Multimedia conference
sessions with support for participants to contribute in text is
another application. Conferences with central support for speech-to-
text conversion is yet another mentioned application.
In all these applications, normally only one participant at a time
will send long text utterances. In some cases, one other participant
will occasionally contribute with a longer comment simultaneously.
That may also happen in some rare cases when text is interpreted to
text in another language in a conference. Apart from these cases,
other participants are only expected to contribute with very brief
utterings while others are sending text.
Text is supposed to be human generated, by some text input means,
such as typing on a keyboard or using speech-to-text technology.
Occasional small cut-and-paste operations may appear even if that is
not the initial purpose of real-time text.
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The real-time characteristics of real-time text is essential for the
participants to be able to contribute to a conversation. If the text
is too much delayed from typing a letter to its presentation, then,
in some conference situations, the opportunity to comment will be
gone and someone else will grab the turn. A delay of more than one
second in such situations is an obstacle for good conversation.
2. Overview over the two specified solutions
This section contains a brief introduction of the two methods
specified in this document.
2.1. Negotiated use of the RFC 4103 format for multi-party transmission
in a single RTP stream
The main purpose of this document is to specify a method for true
multi-party real-time text mixing for multi-party aware endpoints.
The method 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 multi-party situation. It is done by completing a
negotiation for this kind of multi-party capability and by indicating
source in the CSRC element in the RTP packets. Specific
considerations are made to be able to recover text after packet loss.
The detailed procedures for the multi-party aware case are specified
in Section 3
Please use [RFC4103] as reference when reading the specification.
2.2. Mixing for multi-party unaware endpoints
A method is also specified in this document for cases when the
endpoint participating in a multi-party call does not itself
implement any solution for multi-party handling of real-time text.
The solution requires the mixer to insert text dividers and readable
labels and only send text from one source at a time until a suitable
point appears for source change. This solution is a fallback method
with functional limitations. It acts on the presentation level.
A party performing as a mixer, which has not negotiated the "rtt-
mixer" sdp media attribute, but negotiated a "text/red" or "text/
t140" format in a session with a participant SHOULD, if nothing else
is specified for the application, format transmitted text to that
participant to be suitable to present on a multi-party unaware
endpoint as further specified in Section 4.2.
3. Details for the multi-party aware mixing case
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3.1. Offer/answer considerations
RFC 4103[RFC4103] specifies use of RFC 3550 RTP[RFC3550], and a
redundancy format "text/red" for increased robustness of real-time
text transmission. This document updates RFC 4103[RFC4103] by
introducing a capability negotiation for handling multi-party real-
time text, a way to indicate the source of transmitted text, and
rules for efficient timing of the transmissions interleaved from
different sources.
The capability negotiation is based on use of the sdp media attribute
"rtt-mixer".
Both parties shall indicate their capability in a session setup or
modification, and evaluate the capability of the counterpart.
The syntax is as follows:
"a=rtt-mixer"
3.2. Actions depending on capability negotiation result
A transmitting party SHALL send text according to the multi-party
format only when the negotiation for this method was successful and
when the CC field in the RTP packet is set to 1. In all other cases,
the packets SHALL be populated and interpreted as for a two-party
session.
A party which has negotiated the "rtt-mixer" sdp media attribute MUST
populate the CSRC-list and format the packets according to Section 3
if it acts as an rtp-mixer and sends multi-party text.
A party which has negotiated the "rtt-mixer" sdp media attribute MUST
interpret the contents of the "CC" field the CSRC-list and the
packets according to Section 3 in received rtp packets in the
corresponding RTP stream.
A party not performing as a mixer MUST not include the CSRC list.
3.3. Use of fields in the RTP packets
The CC field SHALL show the number of members in the CSRC list, which
SHALL be one (1) in transmissions from a mixer involved in a multi-
party session, and otherwise 0.
When transmitted from a mixer during a multi-party session, a CSRC
list SHALL be included in the packet. The single member in the CSRC-
list SHALL contain the SSRC of the source of the T140blocks in the
packet. When redundancy is used, the recommended level of redundancy
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is to use one primary and two redundant generations of T140blocks.
In some cases, a primary or redundant T140block is empty, but is
still represented by a member in the redundancy header.
From other aspects, the contents of the RTP packets are equal to what
is specified in [RFC4103].
3.4. Initial transmission of a BOM character
As soon as a participant is known to participate in a session with
another entity and being available for text reception, a Unicode BOM
character SHALL be sent to it by the other entity according to the
procedures in this section. If the transmitter is a mixer, then the
source of this character SHALL be indicated to be the mixer itself.
Note that the BOM character SHALL be transmitted with the same
redundancy procedures as any other text.
3.5. Keep-alive
After that, the transmitter SHALL send keep-alive traffic to the
receiver(s) at regular intervals when no other traffic has occurred
during that interval, if that is decided for the actual connection.
Recommendations for keep-alive can be found in [RFC6263].
3.6. Transmission interval
A "text/red" or "text/t140" transmitter in a mixer SHOULD send
packets distributed in time as long as there is something (new or
redundant T140blocks) to transmit. The maximum transmission interval
SHOULD then be 330 ms. It is RECOMMENDED to send next packet to a
receiver as soon as new text to that receiver is available, as long
as the time after the latest sent packet to the same receiver is more
than or equal to 100 ms, and also the maximum character rate ("CPS")
to the receiver is not exceeded. The intention is to keep the
latency low and network load limited while keeping a good protection
against text loss in bursty packet loss conditions.
For a transmitter not acting in a mixer, the same transmission
interval principles apply, but the maximum transmission interval
SHOULD be 300 ms.
3.7. Only one source per packet
New and redundant text from one source SHALL be transmitted in the
same packet if available for transmission at the same time. Text
from different sources MUST NOT be transmitted in the same packet.
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3.8. Do not send received text to the originating source
Text received to a mixer from a participant SHOULD NOT be included in
transmission from the mixer to that participant.
3.9. Clean incoming text
A mixer SHALL handle reception, recovery 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.
3.10. 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
SHOULD 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.
3.11. 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 SHOULD be next in turn to get all its available unsent
text transmitted. The age of redundant text SHOULD then be
considered to be 330 ms after its previous transmission.
3.12. Text placement in packets
The mixer SHOULD compose and transmit an RTP packet to a receiver
when one of the following conditions has occurred:
* 100 ms has passed since the latest transmission to that receiver,
and there is unsent text available for transmission.
* New text has arrived and more than 100 ms has passed since latest
transmission to that receiver.
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* 330 ms has passed since already transmitted text was queued for
transmission as redundant text, and more than 100 ms has passed
since the latest transmission to that receiver, and the redundant
text is still not sent.
At time of transmission, the mixer SHALL populate the RTP packet with
all T140blocks queued for transmission originating from the source in
turn for transmission as long as this is not in conflict with the
allowed number of characters per second ("CPS") or the maximum packet
size. In this way, the latency of the latest received text is kept
low even in moments of simultaneous transmission from many sources.
Redundant text SHALL also be included. See Section 3.14
The SSRC of the source shall be placed as the only member in the
CSRC-list.
Note: The CSRC-list in an RTP packet only includes the participant
who's text is included in text blocks. It is not the same as the
total list of participants in a conference. With audio and video
media, the CSRC-list would often contain all participants who are not
muted whereas text participants that don't type are completely silent
and thus are not represented in RTP packet CSRC-lists.
3.13. Empty T140blocks
If no unsent T140blocks were available for a source at the time of
populating a packet, but T140blocks are available which have not yet
been sent the full intended number of redundant transmissions, then
the primary T140block for that source is composed of an empty
T140block, and populated (without taking up any length) in a packet
for transmission. The corresponding SSRC SHALL be placed as usual in
its place in the CSRC-list.
The first packet in the session, the first after a source switch and
the first after a pause SHALL be poulated with the available
T140blocks for the source in turn to be sent as primary, and empty
T140blocks for the agreed number of redundancy generations.
3.14. Creation of the redundancy
The primary T140block from a source in the latest transmitted packet
is saved for populating the first redundant T140block for that source
in next transmission of text from that source. The first redundant
T140block for that source from the latest transmission is saved for
populating the second redundant T140block in next transmission of
text from that source.
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Usually this is the level of redundancy used. If a higher number of
redundancy is negotiated, then the procedure SHALL be maintained
until all available redundant levels of T140blocks are placed in the
packet. If a receiver has negotiated a lower number of "text/red"
generations, then that level shall be the maximum used by the
transmitter.
The T140blocks saved for transmission as redundant data are assigned
a planned transmission time 330 ms after the current time, but SHOULD
be transmitted earlier if new text for the same source gets in turn
for transmission before that time.
3.15. Timer offset fields
The timestamp offset values are inserted in the redundancy header,
with the time offset from the RTP timestamp in the packet when the
corresponding T140block was sent as primary.
The timestamp offsets are expressed in the same clock tick units as
the RTP timestamp.
The timestamp offset values for empty T140blocks have no relevance
but SHOULD be assigned realistic values.
3.16. Other RTP header fields
The number of members in the CSRC list ( 0 or 1) shall be placed in
the "CC" header field. Only mixers place value 1 in the "CC" field.
A value of "0" indicates that the source is the transmitting device
itself and that the source is indicated by the SSRC field. This
value is used by endpoints, and by mixers sending data that it is
source of itself.
The current time SHALL be inserted in the timestamp.
The SSRC of the mixer for the RTT session SHALL be inserted in the
SSRC field of the RTP header.
The M-bit shall be handled as specified in [RFC4103].
3.17. Pause in transmission
When there is no new T140block to transmit, and no redundant
T140block that has not been retransmitted the intended number of
times from any source, the transmission process can stop until either
new T140blocks arrive, or a keep-alive method calls for transmission
of keep-alive packets.
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3.18. RTCP considerations
A mixer SHALL send RTCP reports with SDES, CNAME and NAME information
about the sources in the multi-party call. This makes it possible
for participants to compose a suitable label for text from each
source.
Integrity considerations SHALL be considered when composing these
fields.
3.19. Reception of multi-party contents
The "text/red" receiver included in an endpoint with presentation
functions will receive RTP packets in the single stream from the
mixer, and SHALL distribute the T140blocks for presentation in
presentation areas for each source. Other receiver roles, such as
gateways or chained mixers are also feasible, and requires
consideration if the stream shall just be forwarded, or distributed
based on the different sources.
3.19.1. Multi-party vs two-party use
If the "CC" field value of a received packet is 1, it indicates that
multi-party transmission is active, and the receiver MUST be prepared
to act on the source according to its role. If the CC value is 0,
the connection is point-to-point.
3.19.2. Level of redundancy
The used level of redundancy generations SHALL be evaluated from the
received packet contents. The number of generations (including the
primary) is equal to the number of members in the redundancy header.
3.19.3. Empty T140blocks
Empty T140blocks are included as fillers for unused primary or
redundancy levels in the packets. They just do not provide any
contents.
3.19.4. Detection and indication of possible text loss
The RTP sequence numbers of the received packets SHALL be monitored
for gaps and packets out of order. If a sequence number gap appears
and still exists after some defined short time for jitter resolution,
the packets in the gap SHALL be regarded lost.
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If it is known that only one source is active in the RTP session,
then it is likely that a gap equal to or larger than the agreed
number of redundancy generations (including the primary) causes text
loss. In that case a t140block SHALL be created with a marker for
possible text loss [T140ad1] and assigned to the source and inserted
in the reception buffer for that source.
If it is known that more than one source is active in the RTP
session, then it is not possible in general to evaluate if text was
lost when packets were lost. With two active sources and the
recommended number of redundancy generations (3), it can take a gap
of five 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. Then
a t140block SHALL be created with a marker for possible text loss
[T140ad1] and assigned to the SSRC of the transmitter as a general
input from the mixer.
Implementations MAY apply more refined methods for more reliable
detection of if text was lost or not. Any refined method SHOULD
rather falsely mark possible loss when there was no loss instead of
not marking possible loss when there was loss.
3.19.5. Extracting text and handling recovery
When applying the following procedures, the effects MUST be
considered of possible timestamp wrap around and the RTP session
possibly changing SSRC.
When a packet is received in an RTP session using the packetization
for multi-party aware endpoints, its T140blocks SHALL be extracted in
the following way. The description is adapted to the default
redundancy case using the original and two redundant generations.
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.
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Note: The normal case is that in the first packet, only the primary
data has contents. The redundant data has contents in the first
received packet from a source only after initial packet loss.
If the packet is not the first packet from a source, then if the
second generation redundant data is available, its timestamp SHALL be
created by subtracting its timestamp offset from the RTP timestamp.
If the resulting timestamp is later than the latest retrieved data
from the same source, then the redundant data SHALL be retrieved and
appended to the receive buffer. The process SHALL be continued in
the same way for the first generation redundant data. After that,
the primary data SHALL be retrieved from the packet and appended to
the receive buffer for the source.
3.19.6. Delete 'BOM'
Unicode character 'BOM' is used as a start indication and sometimes
used as a filler or keep alive by transmission implementations.
These SHALL be deleted after extraction from received packets.
3.20. Performance considerations
This solution has good performance for up to five participants
simultaneously sending text. At higher numbers of participants
simultaneously sending text, a jerkiness is visible in the
presentation of text. With ten participants simultaneously
transmitting text, the jerkiness is about one second. Even so, the
transmission of text catches up, so there is limited delay of new
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.
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3.21. Security for session control and media
Security SHOULD be applied on both session control and media. In
applications where legacy endpoints without security may exist, a
negotiation SHOULD be performed to decide if security by encryption
will be applied. If no other security solution is mandated for the
application, then RFC 8643 OSRTP [RFC8643] SHOULD be applied to
negotiate SRTP media security with DTLS. Most SDP examples below are
for simplicity expressed without the security additions. The
principles (but not all details) for applying DTLS-SRTP [RFC5764]
security is shown in a couple of the following examples.
3.22. SDP offer/answer examples
This sections shows some examples of SDP for session negotiation of
the real-time text media in SIP sessions. Audio is usually provided
in the same session, and sometimes also video. The examples only
show the part of importance for the real-time text media.
Offer example for "text/red" format and multi-party support:
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Answer example from a multi-party capable device
m=text 14000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Offer example for "text/red" format including multi-party
and security:
a=fingerprint: (fingerprint1)
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
The "fingerprint" is sufficient to offer DTLS-SRTP, with the media
line still indicating RTP/AVP.
Note: For brevity, the entire value of the SDP fingerprint attribute
is not shown in this and the following example.
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Answer example from a multi-party capable device with security
a=fingerprint: (fingerprint2)
m=text 16000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
With the "fingerprint" the device acknowledges use of SRTP/DTLS.
Answer example from a multi-party unaware device that also
does not support security:
m=text 12000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
3.23. Packet sequence example from interleaved transmission
This example shows a symbolic flow of packets from a mixer including
loss and recovery. The sequence includes interleaved transmission of
text from two RTT sources A and B. P indicates primary data. R1 is
first redundant generation data and R2 is the second redundant
generation data. A1, B1, A2 etc are text chunks (T140blocks)
received from the respective sources and sent on to the receiver by
the mixer. X indicates 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 area A. The mixer is now assumed to have received text
from source B and will send that text 100 ms after packet 101.
Transmission of A2 and A3 as redundancy is planned for 330 ms after
packet 101 if no new text from A is ready to be sent before that.
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|-----------------------|
|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 dropped in network problems. It
contains redundancy for A. Sending A3 as second level
redundancy is planned for 330 ms after packet 104.
X------------------------|
X Seq no 104, Timer=20820|
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 in 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.
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|------------------------|
| 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 103, and 104 is detected.
Assume that no other loss was detected the last second.
Then it can be concluded that nothing was totally lost.
R2 is checked. Its original time was 21040-660=20400.
A packet with text from A was received with that
timestamp, so nothing needs to be recovered.
B1 and B2 still needs to be transmitted as redundancy.
This is planned 330 ms after packet 105. But that
would be at 21150 which is only 90 ms after the
latest packet. It is instead transmitted at
time 21160.
|-----------------------|
|Seq no 106, Timer=21160|
|CC=1 |
|CSRC list B |
| R2: B1, Offset=660 |
| R1: B2, Offset=340 |
| P: Empty |
|-----------------------|
Packet 106 is received.
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 101 THAT was received. So B1 does not
need to be retrieved. The first level redundancy in packet 106 has
offset 340. The timestamp of packet 106 minus 340 is 20820. That is
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.
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3.24. Maximum character rate "CPS"
The default maximum rate of reception of "text/t140" real-time text
is in RFC 4103 [RFC4103] specified to be 30 characters per second.
The value MAY be modified in the CPS parameter of the FMTP attribute
in the media section for the "text/t140" media. A mixer combining
real-time text from a number of sources may occasionally have a
higher combined flow of text coming from the sources. Endpoints
SHOULD therefore specify a suitable higher value for the CPS
parameter, corresponding to its real reception capability. A value
for "CPS" of 90 is the default for the "text/t140" stream in the
"text/red" format when multi-party real-time text is negotiated. See
RFC 4103 [RFC4103] for the format and use of the CPS parameter. The
same rules apply for the multi-party case except for the default
value.
4. Presentation level considerations
ITU-T T.140 [T140] provides the presentation level requirements for
the RFC 4103 [RFC4103] transport. T.140 [T140] has functions for
erasure and other formatting functions and has the following general
statement for the presentation:
"The display of text from the members of the conversation should be
arranged so that the text from each participant is clearly readable,
and its source and the relative timing of entered text is visualized
in the display. Mechanisms for looking back in the contents from the
current session should be provided. The text should be displayed as
soon as it is received."
Strict application of T.140 [T140] is of essence for the
interoperability of real-time text implementations and to fulfill the
intention that the session participants have the same information of
the text contents of the conversation without necessarily having the
exact same layout of the conversation.
T.140 [T140] specifies a set of presentation control codes to include
in the stream. Some of them are optional. Implementations MUST be
able to ignore optional control codes that they do not support.
There is no strict "message" concept in real-time text. 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 replacement for Line
Separator. The "CRLF" combination SHALL be erased by just one
erasing action, just as the Line Separator. Presentation functions
are allowed to group text for presentation in smaller groups than the
line separators imply and present such groups with source indication
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together with text groups from other sources (see the following
presentation examples). Erasure has no specific limit by any
delimiter in the text stream.
4.1. Presentation by multi-party aware endpoints
A multi-party aware receiving party, presenting real-time text MUST
separate text from different sources and present them in separate
presentation fields. The receiving party MAY separate presentation
of parts of text from a source in readable groups based on other
criteria than line separator and merge these groups in the
presentation area when it benefits the user to most easily find and
read text from the different participants. The criteria MAY e.g. be
a received comma, full stop, or other phrase delimiters, or a long
pause.
When text is received from multiple original sources, the
presentation SHOULD provide a view where text is added in multiple
presentation fields.
If the presentation presents text from different sources in one
common area, the presenting endpoint SHOULD insert text from the
local user ended at suitable points merged with received text to
indicate the relative timing for when the text groups were completed.
In this presentation mode, the receiving endpoint SHALL present the
source of the different groups of text. This presentation style is
called the "chat" style here.
A view of a three-party RTT call in chat style is shown in this
example .
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_________________________________________________
| |^|
|[Alice] Hi, Alice here. |-|
| | |
|[Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris. | |
| I thought you should be here. | |
| | |
|[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|[Bob] And I on Wednesday evening. | |
| | |
|[Alice] Can we meet on Thursday evening? | |
| | |
|[Eve] Yes, definitely. How about 7pm. | |
| at the entrance of the restaurant | |
| Le Lion Blanc? | |
|[Eve] we can have dinner and then take a walk |-|
|______________________________________________|v|
| <Eve-typing> But I need to be back to |^|
| the hotel by 11 because I need |-|
| | |
| <Bob-typing> I wou |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 3: Example of a three-party RTT call presented in chat style
seen at participant 'Alice's endpoint.
Other presentation styles than the chat style may be arranged.
This figure shows how a coordinated column view MAY be presented.
_____________________________________________________________________
| Bob | Eve | Alice |
|____________________|______________________|_______________________|
| | |I will arrive by TGV. |
|My flight is to Orly| |Convenient to the main |
| |Hi all, can we plan |station. |
| |for the seminar? | |
|Eve, will you do | | |
|your presentation on| | |
|Friday? |Yes, Friday at 10. | |
|Fine, wo | |We need to meet befo |
|___________________________________________________________________|
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Figure 4: An example of a coordinated column-view of a three-party
session with entries ordered vertically in approximate time-order.
4.2. Multi-party mixing for multi-party unaware endpoints
When the mixer has indicated multi-party capability by the "rtt-
mixer" sdp attribute in an SDP negotiation, but the multi-party
capability negotiation fails with an endpoint, then the agreed "text/
red" or "text/t140" format SHALL be used and the mixer SHOULD compose
a best-effort presentation of multi-party real-time text in one
stream intended to be presented by an endpoint with no multi-party
awareness.
This presentation format has functional limitations and SHOULD be
used only to enable participation in multi-party calls by legacy
deployed endpoints implementing only RFC 4103 without any multi-party
extensions specified in this document.
The principles and procedures below do not specify any new protocol
elements. They are instead composed from the information in ITU-T
T.140 [T140] and an ambition to provide a best effort presentation on
an endpoint which has functions only for two-party calls.
The mixer mixing for multi-party unaware endpoints SHALL compose a
simulated limited multi-party RTT view suitable for presentation in
one presentation area. The mixer SHALL group text in suitable groups
and prepare for presentation of them by inserting a new line between
them if the transmitted text did not already end with a new line. A
presentable label SHOULD be composed and sent for the source
initially in the session and after each source switch. With this
procedure the time for source switching is depending on the actions
of the users. In order to expedite source switch, a user can for
example end its turn with a new line.
4.2.1. Actions by the mixer at reception from the call participants
When text is received by the mixer from the different participants,
the mixer SHALL recover text from redundancy if any packets are lost.
The mark for lost text [T140ad1] SHOULD be inserted in the stream if
unrecoverable loss appears. Any Unicode "BOM" characters, possibly
used for keep-alive shall be deleted. The time of creation of text
(retrieved from the RTP timestamp) SHALL be stored together with the
received text from each source in queues for transmission to the
recipients.
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4.2.2. Actions by the mixer for transmission to the recipients
The following procedure SHOULD be applied for each recipient of
multi-part text from the mixer.
The text for transmission SHOULD be formatted by the mixer for each
receiving user for presentation in one single presentation area.
Text received from a participant SHOULD NOT be included in
transmission to that participant. When there is text available for
transmission from the mixer to a receiving party from more than one
participant, the mixer SHOULD switch between transmission of text
from the different sources at suitable points in the transmitted
stream.
When switching source, the mixer SHOULD insert a line separator if
the already transmitted text did not end with a new line (line
separator or CRLF). A label SHOULD be composed from information in
the CNAME and NAME fields in RTCP reports from the participant to
have its text transmitted, or from other session information for that
user. The label SHOULD be delimited by suitable characters (e.g. '[
]') and transmitted. The CSRC SHOULD indicate the selected source.
Then text from that selected participant SHOULD be transmitted until
a new suitable point for switching source is reached.
Integrity considerations SHALL be taken when composing the label.
Seeking a suitable point for switching source SHOULD be done when
there is older text waiting for transmission from any party than the
age of the last transmitted text. Suitable points for switching are:
* A completed phrase ended by comma
* A completed sentence
* A new line (line separator or CRLF)
* A long pause (e.g. > 10 seconds) in received text from the
currently transmitted source
* If text from one participant has been transmitted with text from
other sources waiting for transmission for a long time (e.g. > 1
minute) and none of the other suitable points for switching has
occurred, a source switch MAY be forced by the mixer at next word
delimiter, and also if even a word delimiter does not occur within
a time (e.g. 15 seconds) after the scan for word delimiter
started.
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When switching source, the source which has the oldest text in queue
SHOULD be selected to be transmitted. A character display count
SHOULD be maintained for the currently transmitted source, starting
at zero after the label is transmitted for the currently transmitted
source.
The status SHOULD be maintained for the latest control code for
Select Graphic Rendition (SGR) from each source. If there is an SGR
code stored as the status for the current source before the source
switch is done, a reset of SGR shall be sent by the sequence SGR 0
[009B 0000 006D] after the new line and before the new label during a
source switch. See SGR below for an explanation. This transmission
does not influence the display count.
If there is an SGR code stored for the new source after the source
switch, that SGR code SHOULD be transmitted to the recipient before
the label. This transmission does not influence the display count.
4.2.3. Actions on transmission of text
Text from a source sent to the recipient SHOULD increase the display
count by one per transmitted character.
4.2.4. Actions on transmission of control codes
The following control codes specified by T.140 require specific
actions. They SHOULD cause specific considerations in the mixer.
Note that the codes presented here are expressed in UCS-16, while
transmission is made in UTF-8 transform of these codes.
BEL 0007 Bell Alert in session, provides for alerting during an
active session. The display count SHOULD not be altered.
NEW LINE 2028 Line separator. Check and perform a source switch if
appropriate. Increase display count by 1.
CR LF 000D 000A A supported, but not preferred way of requesting a
new line. Check and perform a source switch if appropriate.
Increase display count by 1.
INT ESC 0061 Interrupt (used to initiate mode negotiation
procedure). The display count SHOULD not be altered.
SGR 009B Ps 006D Select graphic rendition. Ps is rendition
parameters specified in ISO 6429. The display count SHOULD not be
altered. The SGR code SHOULD be stored for the current source.
SOS 0098 Start of string, used as a general protocol element
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introducer, followed by a maximum 256 bytes string and the ST.
The display count SHOULD not be altered.
ST 009C String terminator, end of SOS string. The display count
SHOULD not be altered.
ESC 001B Escape - used in control strings. The display count SHOULD
not be altered for the complete escape code.
Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used
for synchronization and keep-alive. SHOULD be deleted from
incoming streams. Shall be sent first after session establishment
to the recipient. The display count shall not be altered.
Missing text mark (U+FFFD) "Replacement character", represented as a
question mark in a rhombus, or if that is not feasible, replaced
by an apostrophe ', marks place in stream of possible text loss.
SHOULD be inserted by the reception procedure in case of
unrecoverable loss of packets. The display count SHOULD be
increased by one when sent as for any other character.
SGR If a control code for selecting graphic rendition (SGR), other
than reset of the graphic rendition (SGR 0) is sent to a
recipient, that control code shall also be stored as status for
the source in the storage for SGR status. If a reset graphic
rendition (SGR 0) originated from a source is sent, then the SGR
status storage for that source shall be cleared. The display
count shall not be increased.
BS (U+0008) Back Space, intended to erase the last entered character
by a source. Erasure by backspace cannot always be performed as
the erasing party intended. If an erasing action erases all text
up to the end of the leading label after a source switch, then the
mixer must not transmit more backspaces. Instead it is
RECOMMENDED that a letter "X" is inserted in the text stream for
each backspace as an indication of the intent to erase more. A
new line is usually coded by a Line Separator, but the character
combination "CRLF" MAY be used instead. Erasure of a new line is
in both cases done by just one erasing action (Backspace). If the
display count has a positive value it is decreased by one when the
BS is sent. If the display count is at zero, it is not altered.
4.2.5. Packet transmission
A mixer transmitting to a multi-party unaware terminal SHOULD send
primary data only from one source per packet. The SSRC SHOULD be the
SSRC of the mixer. The CSRC list SHOULD contain one member and be
the SSRC of the source of the primary data.
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4.2.6. Functional limitations
When a multi-party unaware endpoint presents a conversation in one
display area in a chat style, it inserts source indications for
remote text and local user text as they are merged in completed text
groups. When an endpoint using this layout receives and presents
text mixed for multi-party unaware endpoints, there will be two
levels of source indicators for the received text; one generated by
the mixer and inserted in a label after each source switch, and
another generated by the receiving endpoint and inserted after each
switch between local and remote source in the presentation area.
This will waste display space and look inconsistent to the reader.
New text can be presented only from one source at a time. Switch of
source to be presented takes place at suitable places in the text,
such as end of phrase, end of sentence, line separator and
inactivity. Therefore the time to switch to present waiting text
from other sources may become long and will vary and depend on the
actions of the currently presented source.
Erasure can only be done up to the latest source switch. If a user
tries to erase more text, the erasing actions will be presented as
letter X after the label.
Text loss because of network errors may hit the label between entries
from different parties, causing risk for misunderstanding from which
source a piece of text is.
These facts makes it strongly RECOMMENDED to implement multi-party
awareness in RTT endpoints. The use of the mixing method for multi-
party-unaware endpoints should be left for use with endpoints which
are impossible to upgrade to become multi-party aware.
4.2.7. Example views of presentation on multi-party unaware endpoints
The following pictures are examples of the view on a participant's
display for the multi-party-unaware case.
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_________________________________________________
| Conference | Alice |
|________________________|_________________________|
| |I will arrive by TGV. |
|[Bob]:My flight is to |Convenient to the main |
|Orly. |station. |
|[Eve]:Hi all, can we | |
|plan for the seminar. | |
| | |
|[Bob]:Eve, will you do | |
|your presentation on | |
|Friday? | |
|[Eve]:Yes, Friday at 10.| |
|[Bob]: Fine, wo |We need to meet befo |
|________________________|_________________________|
Figure 5: Alice who has a conference-unaware client is receiving the
multi-party real-time text in a single-stream. This figure shows how
a coordinated column view MAY be presented on Alice's device.
_________________________________________________
| |^|
|[Alice] Hi, Alice here. |-|
| | |
|[mix][Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris | |
| I thought you should be here. | |
| | |
|[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|[mix][Bob] And I on Wednesday evening. | |
| | |
|[Eve] we can have dinner and then walk | |
| | |
|[Eve] But I need to be back to | |
| the hotel by 11 because I need | |
| |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 6: An example of a view of the multi-party unaware
presentation in chat style. Alice is the local user.
5. Relation to Conference Control
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5.1. Use with SIP centralized conferencing framework
The SIP conferencing framework, mainly specified in RFC
4353[RFC4353], RFC 4579[RFC4579] and RFC 4575[RFC4575] is suitable
for coordinating sessions including multi-party RTT. The RTT stream
between the mixer and a participant is one and the same during the
conference. Participants get announced by notifications when
participants are joining or leaving, and further user information may
be provided. The SSRC of the text to expect from joined users MAY be
included in a notification. The notifications MAY be used both for
security purposes and for translation to a label for presentation to
other users.
5.2. Conference control
In managed conferences, control of the real-time text media SHOULD be
provided in the same way as other for media, e.g. for muting and
unmuting by the direction attributes in SDP [RFC4566].
Note that floor control functions may be of value for RTT users as
well as for users of other media in a conference.
6. Gateway Considerations
6.1. Gateway considerations with Textphones (e.g. TTYs).
Multi-party RTT sessions may involve gateways of different kinds.
Gateways involved in setting up sessions SHALL correctly reflect the
multi-party capability or unawareness of the combination of the
gateway and the remote endpoint beyond the gateway.
One case that may occur is a gateway to PSTN for communication with
textphones (e.g. TTYs). Textphones are limited devices with no
multi-party awareness, and it SHOULD therefore be suitable for the
gateway to not indicate multi-party awareness for that case. Another
solution is that the gateway indicates multi-party capability towards
the mixer, and includes the multi-party mixer function for multi-
party unaware endpoints itself. This solution makes it possible to
make adaptations for the functional limitations of the textphone
(TTY).
More information on gateways to textphones (TTYs) is found in RFC
5194[RFC5194]
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6.2. Gateway considerations with WebRTC.
Gateway operation to real-time text in WebRTC may also be required.
In WebRTC, RTT is specified in
[I-D.ietf-mmusic-t140-usage-data-channel].
A multi-party bridge may have functionality for communicating by RTT
both in RTP streams with RTT and WebRTC t140 data channels. Other
configurations may consist of a multi-party bridge with either
technology for RTT transport and a separate gateway for conversion of
the text communication streams between RTP and t140 data channel.
In WebRTC, it is assumed that for a multi-party session, one t140
data channel is established for each source from a gateway or bridge
to each participant. Each participant also has a data channel with
two-way connection with the gateway or bridge.
The t140 channel used both ways is for text from the WebRTC user and
from the bridge or gateway itself to the WebRTC user. The label
parameter of this t140 channel is used as NAME field in RTCP to
participants on the RTP side. The other t140 channels are only for
text from other participants to the WebRTC user.
When a new participant has entered the session with RTP transport of
rtt, a new t140 channel SHOULD be established to WebRTC users with
the label parameter composed from the NAME field in RTCP on the RTP
side.
When a new participant has entered the multi-party session with RTT
transport in a WebRTC t140 data channel, the new participant SHOULD
be announced by a notification to RTP users. The label parameter
from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP
side, or other available session information.
7. Updates to RFC 4103
This document updates RFC 4103[RFC4103] by introducing an sdp media
attribute "rtt-mixer" for negotiation of multi-party mixing
capability with the [RFC4103] format, and by specifying the rules for
packets when multi-party capability is negotiated and in use.
8. Congestion considerations
The congestion considerations and recommended actions from RFC 4103
[RFC4103] are valid also in multi-party situations.
The first action in case of congestion SHOULD be to temporarily
increase the transmission interval up to two seconds.
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If the unlikely situation appears that more than 20 participants in a
conference send text simultaneously, it will take more than 7 seconds
between presentation of text from each of these participants. More
time than that can cause confusion in the session. It is therefore
RECOMMENDED that the mixer discards such text in excess inserts a
general indication of possible text loss [T140ad1] in the session.
If the main text contributor is indicated in any way, the mixer MAY
avoid deleting text from that participant.
9. Acknowledgements
James Hamlin for format and performance aspects.
10. IANA Considerations
10.1. Registration of the "rtt-mixer" sdp media attribute
[RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
RFC number of this document.]
IANA is asked to register the new sdp attribute "rtt-mixer".
Contact name: IESG
Contact email: iesg@ietf.org
Attribute name: rtt-mixer
Attribute semantics: See RFCXXXX Section 3.1
Attribute value: none
Usage level: media
Purpose: Indicate support by mixer and endpoint of multi-party
mixing for real-time text transmission, using a common RTP-stream
for transmission of text from a number of sources mixed with one
source at a time and the source indicated in a single CSRC-list
member.
Charset Dependent: no
O/A procedure: See RFCXXXX Section 3.1
Mux Category: normal
Reference: RFCXXXX
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11. Security Considerations
The RTP-mixer model requires the mixer to be allowed to decrypt, pack
and encrypt secured text from the conference participants. Therefore
the mixer needs to be trusted. This is similar to the situation for
central mixers of audio and video.
The requirement to transfer information about the user in RTCP
reports in SDES, CNAME and NAME fields, and in conference
notifications, for creation of labels may have privacy concerns as
already stated in RFC 3550 [RFC3550], and may be restricted of
privacy reasons. The receiving user will then get a more symbolic
label for the source.
Participants with malicious intentions may appear and e.g. disturb
the multi-party session by a continuous flow of text, or masquerading
as text from other participants. Counteractions should be to require
secure signaling, media and authentication, and to provide higher
level conference functions e.g. for blocking and expelling
participants.
12. Change history
12.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-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.
12.2. 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
12.3. 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.
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Moved conference control to own section.
Improved clarification of recovery and loss in the packet sequence
example.
A number of editorial corrections and improvements.
12.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08
Deleted the method requiring a new packet format "text/rex" because
of the longer standardization and implementation period it needs.
Focus on use of RFC 4103 text/red format with shorter transmission
interval, and source indicated in CSRC.
12.5. 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..
12.6. 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.
12.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05
nomenclature and editorial improvements
"this document" used consistently to refer to this document.
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12.8. 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.
12.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03
Mention possible need to mute and raise hands as for other media.
---done ----
Make sure that use in two-party calls is also possible and explained.
- may need more wording -
Clarify the RTT is often used together with other media. --done--
Tell that text mixing is N-1. A users own text is not received in
the mix. -done-
In 3. correct the interval to: A "text/rex" transmitter SHOULD send
packets distributed in time as long as there is something (new or
redundant T140blocks) to transmit. The maximum transmission interval
SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a
receiver as soon as new text to that receiver is available, as long
as the time after the latest sent packet to the same receiver is more
than 150 ms, and also the maximum character rate to the receiver is
not exceeded. The intention is to keep the latency low while keeping
a good protection against text loss in bursty packet loss conditions.
-done-
In 1.3 say that the format is used both ways. -done-
In 13.1 change presentation area to presentation field so that reader
does not think it shall be totally separated. -done-
In Performance and intro, tell the performance in number of
simultaneous sending users and introduced delay 16, 150 vs
requirements 5 vs 500. -done --
Clarify redundancy level per connection. -done-
Timestamp also for the last data header. To make it possible for all
text to have time offset as for transmission from the source. Make
that header equal to the others. -done-
Mixer always use the CSRC list, even for its own BOM. -done-
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Combine all talk about transmission interval (300 ms vs when text has
arrived) in section 3 in one paragraph or close to each other. -done-
Documents the goal of good performance with low delay for 5
simultaneous typers in the introduction. -done-
Describe better that only primary text shall be sent on to receivers.
Redundancy and loss must be resolved by the mixer. -done-
12.10. 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.
12.11. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01
2,5,6 More efficient format "text/rex" introduced and attribute
a=rtt-mix deleted.
3. Brief about use of OSRTP for security included- More needed.
4. Brief motivation for the solution and why not rtp-translator is
used added to intro.
7. More limitations for the multi-party unaware mixing method
inserted.
8. Updates to RFC 4102 and 4103 more clearly expressed.
9. Gateway to WebRTC started. More needed.
12.12. 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.
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12.13. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02
to -03
Changed company and e-mail of the author.
Changed title to "RTP-mixer formatting of multi-party Real-time text"
to better match contents.
Check and modification where needed of use of RFC 2119 words SHALL
etc.
More about the CC value in sections on transmitters and receivers so
that 1-to-1 sessions do not use the mixer format.
Enhanced section on presentation for multi-party-unaware endpoints
A paragraph recommending CPS=150 inserted in the performance section.
12.14. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01
to -02
In Abstract and 1. Introduction: Introduced wording about regulatory
requirements.
In section 5: The transmission interval is decreased to 100 ms when
there is text from more than one source to transmit.
In section 11 about SDP negotiation, a SHOULD-requirement is
introduced that the mixer should make a mix for multi-party unaware
endpoints if the negotiation is not successful. And a reference to a
later chapter about it.
The presentation considerations chapter 14 is extended with more
information about presentation on multi-party aware endpoints, and a
new section on the multi-party unaware mixing with low functionality
but SHOULD a be implemented in mixers. Presentation examples are
added.
A short chapter 15 on gateway considerations is introduced.
Clarification about the text/t140 format included in chapter 10.
This sentence added to the chapter 10 about use without redundancy.
"The text/red format SHOULD be used unless some other protection
against packet loss is utilized, for example a reliable network or
transport."
Note about deviation from RFC 2198 added in chapter 4.
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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.
12.15. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00
to -01
Editorial cleanup.
Changed capability indication from fmtp-parameter to SDP attribute
"rtt-mix".
Swapped order of redundancy elements in the example to match reality.
Increased the SDP negotiation section
13. References
13.1. Normative References
[I-D.ietf-mmusic-t140-usage-data-channel]
Holmberg, C. and G. Hellstrom, "T.140 Real-time Text
Conversation over WebRTC Data Channels", Work in Progress,
Internet-Draft, draft-ietf-mmusic-t140-usage-data-channel-
14, 10 April 2020, <https://tools.ietf.org/html/draft-
ietf-mmusic-t140-usage-data-channel-14>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type",
RFC 4102, DOI 10.17487/RFC4102, June 2005,
<https://www.rfc-editor.org/info/rfc4102>.
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[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011,
<https://www.rfc-editor.org/info/rfc6263>.
[RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T.
Stach, "An Opportunistic Approach for Secure Real-time
Transport Protocol (OSRTP)", RFC 8643,
DOI 10.17487/RFC8643, August 2019,
<https://www.rfc-editor.org/info/rfc8643>.
[T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for
multimedia application text conversation", February 1998,
<https://www.itu.int/rec/T-REC-T.140-199802-I/en>.
[T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000),
Protocol for multimedia application text conversation",
February 2000,
<https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>.
13.2. Informative References
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353,
DOI 10.17487/RFC4353, February 2006,
<https://www.rfc-editor.org/info/rfc4353>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <https://www.rfc-editor.org/info/rfc4575>.
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[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents",
BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006,
<https://www.rfc-editor.org/info/rfc4579>.
[RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real-
Time Text over IP Using the Session Initiation Protocol
(SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008,
<https://www.rfc-editor.org/info/rfc5194>.
Author's Address
Gunnar Hellstrom
Gunnar Hellstrom Accessible Communication
SE-13670 Vendelso
Sweden
Email: gunnar.hellstrom@ghaccess.se
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