Real-time text solutions for multi-party sessions
draft-hellstrom-avtcore-multi-party-rtt-solutions-06
Internet Engineering Task Force G. Hellstrom
Internet-Draft Gunnar Hellstrom Accessible Communication
Intended status: Informational 17 December 2020
Expires: 20 June 2021
Real-time text solutions for multi-party sessions
draft-hellstrom-avtcore-multi-party-rtt-solutions-06
Abstract
This document specifies methods for Real-Time Text (RTT) media
handling in multi-party calls. The main discussed transport is to
carry Real-Time text by the RTP protocol in a time-sampled mode
according to RFC 4103. The mechanisms enable the receiving
application to present the received real-time text media, separated
per source, in different ways according to user preferences. Some
presentation related features are also described explaining suitable
variations of transmission and presentation of text.
Call control features are described for the SIP environment. A
number of alternative methods for providing the multi-party
negotiation, transmission and presentation are discussed and a
recommendation for the main ones is provided. The main solution for
SIP based centralized multi-party handling of real-time text is
achieved through a media control unit coordinating multiple RTP text
streams into one RTP stream.
Alternative methods using a single RTP stream and source
identification inline in the text stream are also described, one of
them being provided as a lower functionality fallback method for
endpoints with no multi-party awareness for RTT.
Bridging methods where the text stream is carried without the
contents being dealt with in detail by the bridge are also discussed.
Brief information is also provided for multi-party RTT in the WebRTC
environment.
The intention is to provide background for decisions, specification
and implementation of selected methods.
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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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Requirements Language . . . . . . . . . . . . . . . . . . 5
2. Centralized conference model . . . . . . . . . . . . . . . . 5
3. Requirements on multi-party RTT . . . . . . . . . . . . . . . 6
3.1. General requirements . . . . . . . . . . . . . . . . . . 6
3.2. Performance requirements . . . . . . . . . . . . . . . . 7
4. RTP based solutions . . . . . . . . . . . . . . . . . . . . . 8
4.1. Coordination of text RTP streams . . . . . . . . . . . . 8
4.1.1. RTP-based solutions with a central mixer . . . . . . 8
4.1.1.1. RTP Mixer using default RFC 4103 methods . . . . 9
4.1.1.2. RTP Mixer using the default method but decreased
transmission interval . . . . . . . . . . . . . . . 9
4.1.1.3. RTP Mixer with frequent transmission and indicating
sources in CSRC-list . . . . . . . . . . . . . . . 10
4.1.1.4. RTP Mixer interleaving packets, receiver using
timestamp to recover from loss . . . . . . . . . . 12
4.1.1.5. RTP Mixer with multiple primary data in each packet
and individual sequence numbers . . . . . . . . . . 13
4.1.1.6. RTP Mixer with multiple primary data in each
packet . . . . . . . . . . . . . . . . . . . . . . 14
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4.1.1.7. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy
in the packets . . . . . . . . . . . . . . . . . . 15
4.1.1.8. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy
and separate sequence number in the packets . . . . 17
4.1.1.9. RTP Mixer indicating participants by a control code
in the stream . . . . . . . . . . . . . . . . . . . 19
4.1.1.10. Mixing for multi-party unaware user agents . . . 20
4.1.2. RTP-based bridging with minor RTT media contents
reformatting by the bridge . . . . . . . . . . . . . 22
4.1.2.1. One RTP stream for RTT per participant from the
mixer . . . . . . . . . . . . . . . . . . . . . . . 22
4.1.2.2. Selective Forwarding Middlebox . . . . . . . . . 25
4.1.2.3. Distributing packets in an end-to-end encryption
structure . . . . . . . . . . . . . . . . . . . . . 27
4.1.2.4. Mesh of RTP endpoints . . . . . . . . . . . . . . 27
4.1.2.5. Multiple RTP sessions, one for each
participant . . . . . . . . . . . . . . . . . . . . 28
5. Preferred RTP-based multi-party RTT transport method . . . . 28
6. Session control of RTP-based multi-party RTT sessions . . . . 29
6.1. Implicit RTT multi-party capability indication . . . . . 29
6.2. RTT multi-party capability declared by SIP media-tags . . 31
6.3. SDP media attribute for RTT multi-party capability
indication . . . . . . . . . . . . . . . . . . . . . . . 32
6.4. Simplified SDP media attribute for RTT multi-party
capability indication . . . . . . . . . . . . . . . . . . 33
6.5. SDP format parameter for RTT multi-party capability
indication . . . . . . . . . . . . . . . . . . . . . . . 34
6.6. A text media subtype for support of multi-party rtt . . . 35
6.7. Preferred capability declaration method for RTP-based
transport. . . . . . . . . . . . . . . . . . . . . . . . 35
6.8. Identification of the source of text for RTP-based
solutions . . . . . . . . . . . . . . . . . . . . . . . . 36
7. RTT bridging in WebRTC . . . . . . . . . . . . . . . . . . . 36
7.1. RTT bridging in WebRTC with one data channel per
source . . . . . . . . . . . . . . . . . . . . . . . . . 36
7.2. RTT bridging in WebRTC with one common data channel . . . 37
7.3. Preferred rtt multi-party method for WebRTC . . . . . . . 38
8. Presentation of multi-party text . . . . . . . . . . . . . . 38
8.1. Associating identities with text streams . . . . . . . . 38
8.2. Presentation details for multi-party aware endpoints. . . 39
8.2.1. Bubble style presentation . . . . . . . . . . . . . . 39
8.2.2. Other presentation styles . . . . . . . . . . . . . . 41
9. Presentation details for multi-party unaware endpoints. . . . 41
10. Security Considerations . . . . . . . . . . . . . . . . . . . 41
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42
12. Congestion considerations . . . . . . . . . . . . . . . . . . 42
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 42
14. Change history . . . . . . . . . . . . . . . . . . . . . . . 42
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14.1. Changes to
draft-hellstrom-avtcore-multi-party-rtt-solutions-06 . . 42
14.2. Changes to
draft-hellstrom-avtcore-multi-party-rtt-solutions-05 . . 42
14.3. Changes to
draft-hellstrom-avtcore-multi-party-rtt-solutions-04 . . 42
14.4. Changes to
draft-hellstrom-avtcore-multi-party-rtt-solutions-03 . . 43
14.5. Changes to
draft-hellstrom-avtcore-multi-party-rtt-solutions-02 . . 43
14.6. Changes to
draft-hellstrom-avtcore-multi-party-rtt-solutions-01 . . 43
14.7. Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to
draft-hellstrom-avtcore-multi-party-rtt-solutions-00 . . 43
14.8. Changes from version
draft-hellstrom-mmusic-multi-party-rtt-01 to -02 . . . . 44
15. References . . . . . . . . . . . . . . . . . . . . . . . . . 44
15.1. Normative References . . . . . . . . . . . . . . . . . . 44
15.2. Informative References . . . . . . . . . . . . . . . . . 44
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 47
1. Introduction
Real-time text (RTT) is a medium in real-time conversational
sessions. Text entered by participants in a session is transmitted
in a time-sampled fashion, so that no specific user action is needed
to cause transmission. This gives a direct flow of text in the rate
it is created, that is suitable in a real-time conversational
setting. The real-time text medium can be combined with other media
in multimedia sessions.
Media from a number of multimedia session participants can be
combined in a multi-party session. The present document specifies
how the real-time text streams can be handled in multi-party
sessions. Recommendations are provided for preferred methods.
The description is mainly focused on the transport level, but also
describes a few session and presentation level aspects.
Transport of real-time text is specified in RFC 4103 [RFC4103] RTP
Payload for text conversation. It makes use of RFC 3550 [RFC3550]
Real Time Protocol, for transport. Robustness against network
transmission problems is normally achieved through redundant
transmission based on the principle from RFC 2198 [RFC2198], with one
primary and two redundant transmission of each text element. Primary
and redundant transmissions are combined in packets and described by
a redundancy header. This transport is usually used in the SIP
Session Initiation Protocol RFC 3261 [RFC3261] environment.
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A very brief overview of functions for real-time text handling in
multi-party sessions is described in RFC 4597 [RFC4597] Conferencing
Scenarios, sections 4.8 and 4.10. The present specification builds
on that description and indicates which protocol mechanisms should be
used to implement multi-party handling of real-time text.
Real-time text can also be transported in the WebRTC environment, by
using WebRTC data channels according to RFC-to-be 8865
[I-D.ietf-mmusic-t140-usage-data-channel]. Multi-party aspects for
WebRTC solutions are briefly covered.
1.1. Requirements Language
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 RFC 2119 [RFC2119].
2. Centralized conference model
In the centralized conference model for SIP, introduced in RFC 4353
[RFC4353] "A Framework for Conferencing with the Session Initiation
Protocol (SIP)", one function co-ordinates the communication with
participants in the multi-party session. This function also controls
media mixer functions for the media appearing in the session. The
central function is common for control of all media, while the media
mixers may work differently for each media.
The central function is called the Focus UA. Many variants exist for
setting up sessions including the multipoint control centre. It is
not within scope of this description to describe these, but rather
the media specific handling in the mixer required to handle multi-
party calls with RTT.
The main principle for handling real-time text media in a centralized
conference is that one RTP session for real-time text is established
including the multipoint media control centre and the participating
endpoints which are going to have real-time text exchange with the
others.
The different possible mechanisms for mixing and transporting RTT
differs in the way they multiplex the text streams and how they
identify the sources of the streams. RFC 7667 [RFC7667] describes a
number of possible use cases for RTP. This specification refers to
different sections of RFC 7667 for further reading of the situations
caused by the different possible design choices.
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The recommended method for using RTP based RTT in a centralized
conference model is specified in
[I-D.ietf-avtcore-multi-party-rtt-mix] based on the recommendations
in this document.
Real-time text can also be transported in the WebRTC environment, by
using WebRTC data channels according to
[I-D.ietf-mmusic-t140-usage-data-channel]. Ways to handle multi-
party calls in that environmnent are also specified.
3. Requirements on multi-party RTT
3.1. General requirements
The following general requirements are placed on multi-party RTT:
A solution shall be applicable to IMS (3GPP TS 22.173)[TS22173],
SIP based VoIP and Next Generation Emergency Services (NENA i3
[NENAi3], ETSI TS 103 479 [TS103479], RFC 6443[RFC6443]).
The transmission interval for text should not be longer than 500
milliseconds when there is anything available to send. Ref ITU-T
T.140 [T140].
If text loss is detected or suspected, a missing text marker
should be inserted in the text stream. Ref ITU-T T.140 Amendment
1 [T140ad1]. ETSI EN 301 549 [EN301549]
The display of text from the members of the conversation shall 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. Ref ITU-T T.140
[T140]
Bridges must be multimedia capable (voice, video, text). Ref NENA
i3 STA-010.2. [NENAi3]
It MUST be possible to use real-time text in conferences both as a
medium of discussion between individual participants (for example,
for sidebar discussions in real-time text while listening to the
main conference audio) and for central support of the conference
with real-time text interpretation of speech. Ref (R7) in RFC
5194.[RFC5194]
It should be possible to protect RTT contents with usual means for
privacy and integrity. Ref RFC 6881 section 16. [RFC6881]
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Conferencing procedures are documented in RFC 4579 [RFC4579]. Ref
NENA i3 STA-010.2.[NENAi3]
Conferencing applies to any kind of media stream by which users
may want to communicate. Ref 3GPP TS 24.147 [TS24147]
The framework for SIP conferences is specified in RFC 4353
[RFC4353]. Ref 3GPP TS 24.147 [TS24147]
3.2. Performance requirements
The mixer performance requirements can be expressed in one number,
extracted from the user requirements on real-time text expressed in
ITU-T F.700, where it is stated that for "good" usability, text
characters should not be delayed more than 1 second from creation to
presentation. For "usable" usability the figure is 2 seconds. The
main factor behind these limits is from when taking turns in a
conversation gets disturbed by a delay of when a response gets
visible to the receiving part. If that times get too long, the
receiving part gets unsure if the previous utterance was well
perceived and the receiving part maybe prepares for repetition. This
is similar to the same effect in voice communication, where the
usability limit is 400 ms delay.
Another important factor in a multi-party conference is the
opportunity for a participant using real-time text to provide timely
comments and get a chance to enter the discussion if the majority of
participants use voice in the conference. A complicating factor when
stating the requirements is that some transport methods do not cause
a total delay, but instead an increasing jerkiness when the number of
simultaneously sending participants is increased.
It should however be remembered that the expected number of
participants sending real-time text simultaneously is low. Just as
with voice or sign language, the capability of the participants to
perceive utterances from more than one participant at a time is very
limited. Therefore the normal case in multi-party situations is that
one participant at a time is the main provider of text. Others might
usually just provide very brief comments such as "yes" or "no" or
"may I comment?". Only at very rare situations two participants
provide more information simultaneously.
* The number of expected simultaneously transmitting users is
different for different applications. In all cases, just one
transmitting user is the normal case. Two simultaneously
transmitting participants can occasionally be expected in
emergency services, relay services, small unmanaged conferences
and group calls and large managed conferences. Three
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simultaneously transmitting participants may appear occasionally
in large unmanaged conferences. The following can therefore
express the performance requirement.
* The mean delay of text passing the mixer introduced when only one
participant is sending text should be kept to a minimum and should
not be more than 400 ms.
* The mean delay of text passing the mixer should not be more than 1
second during moments when up to three users are sending text
simultaneously.
* For the very rare case that more than three participants send text
simultaneously, the mixer may take action to limit the introduced
delay of the text passing the mixer to 7 seconds e.g. by
discarding text from some participants and instead inserting a
general warning about possible text loss in the stream.
* The load on network and nodes should be limited. This is usually
achieved by setting a limit for how many packets per second that
may be sent from a mixer to each participant. While two-party use
by RFC 4103, limits the load to 3.3 packets per second, a
realistic limit for mixers could be 10 packets per second. This
is still just a small fraction of what is commonly transmitted in
real-time video and audio, so in known environments it may be
possible to increase the packet rate if needed to keep latency
low.
4. RTP based solutions
4.1. Coordination of text RTP streams
Coordinating and sending text RTP streams in the multi-party session
can be done in a number of ways. The most suitable methods are
specified here with pros and cons.
A receiving and presenting endpoint MUST separate text from the
different sources and identify and display them accordingly.
4.1.1. RTP-based solutions with a central mixer
A set of solutions can be based on the central RTP mixer. They are
described here and a preferred method selected.
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4.1.1.1. RTP Mixer using default RFC 4103 methods
Without any extra specifications, a mixer would transmit with 300
milliseconds intervals, and use RFC 4103 [RFC4103] with the default
redundancy of one original and two redundant transmissions. The
source of the text would be indicated by a single member in the CSRC
list. Text from different sources cannot be transmitted in the same
packet. Therefore, from the time when the mixer sent one piece of
new text from one source, it will need to transmit that text again
twice as redundant data, before it can send text from another source.
The jerkiness = time between transmission of new text is 900 ms.
This is clearly insufficient.
Pros:
Only a capability negotiation method is needed. No other update of
standards are needed, just a general remark that traditional RTP-
mixing is used.
Cons:
Clearly insufficient mixer switching performance.
A bit complex handling of transmission when there is new text
available from more than one source. The mixer needs to send two
packets more with redundant text from the current source before
starting to send anything from the other source.
4.1.1.2. RTP Mixer using the default method but decreased transmission
interval
This method makes use of the default RTP-mixing method briefly
described in Section 4.1.1.1. The only difference is that the
transmission interval is decreased to 100 milliseconds when there is
text from more than one source available for transmission. The
jerkiness is 300 ms. The mean delay with two simultaneously sending
participants is 250 ms, and with three simultaneously sending
participants 500 ms. This is acceptable performance.
Pros:
Minor influence on standards
Can be relatively rapidly be introduced in the intended technical
environments.
Can be declared in sdp as the already existing "text/red" format with
a multi-party attribute for capability negotiation.
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Cons:
The introduced jerkiness of new text from more than the required
three simultaneously sending sources is high.
Slightly higher risk for loss of text at bursty packet loss than for
the recommended transmission interval (300 ms) for RFC 4103.
When complete loss of packets occur (beyond recovery), it is not
possible to deduce from which source text was lost.
A bit complex handling of transmission when there is new text
available from more than one source. The mixer needs to send two
packets more with redundant text from the current source before
starting to send anything from the other source.
4.1.1.3. RTP Mixer with frequent transmission and indicating sources in
CSRC-list
An RTP media mixer combines text from participants into one RTP
stream, thus all using the same destination address/port combination,
the same RTP SSRC, and one sequence number series as described in
Section 7.1 and 7.3 of RTP RFC 3550 [RFC3550] about the Mixer
function. This method is also briefly described in RFC 7667, section
3.6.1 Media mixing mixer [RFC7667].
The sources of the text in each RTP packet are identified by the CSRC
list in the RTP packets, containing the SSRC of the initial sources
of text. The order of the CSRC parameters is with the SSRC of the
source of the primary text first, followed by the SSRC of the first
level redundancy, and then the second level redundancy.
The transmission interval should be 100 milliseconds when there is
text to transmit from more than one source, and otherwise 300 ms.
The identification of the sources is made through the CSRC fields and
can be made more readable at the receiver through the RTCP SDES CNAME
and NAME packets as described in RTP[RFC3550].
Information provided through the notification according to RFC 4575
[RFC4575] when the participant joined the conference provides also
suitable information and a reference to the SSRC.
A receiving endpoint is supposed to separate text items from the
different sources and identify and display them accordingly.
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The ordered CSRC lists in the RFC 4103 [RFC4103] packets make it
possible to recover from loss of one and two packets in sequence and
assign the recovered text to the right source. For more loss, a
marker for possible loss should be inserted or presented.
The conference server needs to have authority to decrypt the payload
in the received RTP packets in order to be able to recover text from
redundant data or insert the missing text marker in the stream, and
repack the text in new packets.
Even if the format is very similar to "text/red" of RFC 4103, it
needs to be declared as a new media subtype, e.g. "text/rex".
Pros:
This method has low overhead and less complexity than the methods in
Section 4.1.1.1, Section 4.1.1.2, Section 4.1.1.4 and
Section 4.1.1.6.
When loss of packets occur, it is possible to recover text from
redundancy at loss of up to the number of redundancy levels carried
in the RFC 4103 [RFC4103] stream (normally primary and two redundant
levels).
This method can be implemented with most RTP implementations.
The source switching performance is sufficient for well-behaving
conference participants. The jerkiness is 100 ms.
Cons:
When more consecutive packet loss than the number of generations of
redundant data appears, it is not possible to deduce the sources of
the totally lost data.
Slightly higher risk for loss of text at bursty packet loss than for
the recommended transmission interval for RFC 4103.
Requires a different sub media format, e.g. "text/rex". This takes a
long time in standardisation and releases of target technical
environments.
The conference server needs to be allowed to decrypt/encrypt the
packet payload. This is however normal for media mixers for other
media.
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4.1.1.4. RTP Mixer interleaving packets, receiver using timestamp to
recover from loss
This method has text only from one source per packet, as the original
RFC 4103 [RFC4103] specifies. Packets with text from different
sources are instead allowed to be interleaved. The recovery
procedure in the receiver makes use of the RTP timestamp and
timestamp offsets in the redundancy headers to evaluate if a piece of
redundant data was received earlier or not as a base for decision if
the redundant data should be recovered or not in case of packet loss.
In this method, the transmission interval is 100 milliseconds when
text (new or redundant) from more than one source is available for
transmission. Otherwise it is 320 ms or following the timing of
received packets.
Pros:
The format of each packet is equal to what is specified in RFC 4103
[RFC4103].
The source switching performance is sufficient and good. Text from
five participants can be transmitted simultaneously with 500
milliseconds interval per source.
New text from five simultaneous sources can be transmitted within 500
milliseconds. This is sufficient.
Recovery from packet loss with five simultaneous sources takes 1
second. This is good and implies good protextion against bursty
packet loss causing resulting text loss.
Cons:
The recovery time in case of packet loss is long with more than ten
simultaneously sending participants. Then it will be more than 2
seconds.
The recovery procedure is different from what is described in RFC
4103 [RFC4103].
It will in many cases of loss of multiple packets not be possible to
deduce if there was any resulting loss of text. A mark for possible
loss should be inserted in cases when there might have been resulting
loss.
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4.1.1.5. RTP Mixer with multiple primary data in each packet and
individual sequence numbers
This method allows primary as well as redundant text from more than
one source per packet. The packet payload contains an ordered set of
redundant and primary data with the same number of generations of
redundancy as once agreed in the SDP negotiation. The data header
reflects these parts of the payload. The CSRC list contains one CSRC
member per source in the payload and in the same order. An
individual sequence number per source is included in the data header
replacing the t140 payload type number that is instead assumed to be
constant in this format. This allows an individual extra sequence
number per source with maximum value 127, suitable for checking for
which source loss of text appeared when recovery was not possible.
The data header would contain the following fields:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Source-seq | timestamp offset | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Where "Source-seq" is the sequence number per source.
The maximum number of members in the CSRC-list is 15, and that is
therefore the maximum number of sources that can be represented in
each packet provided that all data can be fitted into the size
allowable in one packet.
Transmission is done as soon as there is new text available, but not
with shorter interval than 150 ms and not longer than 300 ms while
there is anything to send.
A new media subtype is needed, e.g. "text/rex".
This is an SDP offer example for both traditional "text/red"
and multi-party "text/rex" format:
m=text 11000 RTP/AVP 101 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=rtpmap:101 rex/1000
a=fmtp:100 98/98/98
a=fmtp:101 98/98/98
Pros:
The source switching performance is good. Text from 15 participants
can be transmitted simultaneously.
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New text from 15 simultaneous sources can be transmitted within 300
milliseconds. This is good performance.
When more consecutive packet loss than the number of generations of
redundant data appears, it is still possible to deduce the sources of
the totally lost data, when next text from these sources arrive.
Cons:
The format of each packet is different from what is specified in RFC
4103 [RFC4103].
The processing time in standard organisation will be long.
A new media subtype is needed, causing a bit complex negotiation.
The recovery procedure is a bit complex.
4.1.1.6. RTP Mixer with multiple primary data in each packet
This method allows primary as well as redundant text from more than
one source per packet. The packet payload contains an ordered set of
redundant and primary data with the same number of generations of
redundancy as once agreed in the SDP negotiation. The data header
reflects these parts of the payload. The CSRC list contains one CSRC
member per source in the payload and in the same order.
The maximum number of members in the CSRC-list is 15, and that is
therefore the maximum number of sources that can be represented in
each packet provided that all data can be fitted into the size
allowable in one packet.
Transmission is done as soon as there is new text available, but not
with shorter interval than 150 ms and not longer than 300 ms while
there is anything to send.
A new media subtype is needed, e.g. "text/rex".
SDP would be the same as in Section 4.1.1.6.
Pros:
The source switching performance is good. Text from 15 participants
can be transmitted simultaneously.
New text from 15 simultaneous sources can be transmitted within 150
milliseconds. This is good performance.
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Cons:
The format of each packet is different from what is specified in RFC
4103 [RFC4103].
A new media subtype is needed.
A new media subtype is needed, causing a bit complex negotiation.
The processing time in standard organisation will be long.
The recovery procedure is a bit complex [RFC4103].
When more consecutive packet loss than the number of generations of
redundant data appears, it is not possible to deduce the sources of
the totally lost data.
4.1.1.7. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy in the
packets
This method allows primary data from one source and redundant text
from other sources in each packet. The packet payload contains
primary data in "text/t140" format, and redundant data in RFC 5109
FEC [RFC5109] format called "text/ulpfec". That means that the
redundant data contains the sequence number and the CSRC and other
characteristics from the RTP header when the data was sent as
primary. The redundancy can be sent at a selected number of packets
after when it was sent as primary, in order to improve the protection
against bursty packet loss. The redundancy level is recommended to
be the same as in original RFC 4103.
RFC 4103 says that the protection against loss can be made by other
methods than plain redundancy, so this method is in line with that
statement.
Transmission is done as soon as there is new text available, but not
with shorter interval than 100 ms and not longer than 300 ms while
there is anything to send (new or redundant text).
When more consecutive packet loss than the number of generations of
redundant data appears, it is not possible to deduce the sources of
the totally lost data.
The sdp can indicate the format as "text/red" with "text/ulpfec"
redundant data in this way. with traditional RFC 4103 with "text/red"
with "text/t140" as redundant data as a fallback.
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m=text 49170 RTP/AVP 98 101 100 102
a=rtpmap:98 red/1000
a=fmtp:98 100/102/102
a=rtpmap:102 ulpfec/1000
a=rtpmap:100 t140/1000
a=rtpmap:101 red/1000
a=fmtp:101 100/100/100
a=fmtp:100 cps=200
The "text/ulpfec" format includes an indication of how far back the
redundancy belongs, making it possible to cover bursty packet loss
better than the other formats with short transmission intervals. For
real-time text, it is recommended to send three packets between the
primary and the redundant transmissions of text. That makes the
transmission cover between 500 and 1500 ms of bursty packet loss.
The variation is because of the varying packet interval between many
and one simultaneously transmitting source.
The "text/ulpfec" format has a number of parameters. One is the
length of the data to be protected which in this case must be the
whole t140block.
Pros:
The source switching performance is good. Text from 5 participants
can be transmitted within 500 ms.
Good recovery from bursty packet loss.
The method is based on existing standards. No new registrations are
needed.
Cons:
When more consecutive packet loss than the number of generations of
redundant data appears, it is not possible to deduce the sources of
the totally lost data.
Even if the switching performance is good, it is not as good as for
the method called "RTP Mixer with multiple primary data in each
packet "Section 4.1.1.6. With more than 5 simultaneously sending
sources, there will be a noticeable delay of text of over 500 ms,
with 100 ms added per simultaneous source. This is however beyond
the requirements and would be a concern only in congestion
situations.
The recovery procedure is a bit complex [RFC5109].
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There is more overhead in terms of extra data and extra packets sent
than in the other methods. With the recommended two redundant
generations of data, each packet will be 36 bytes longer than with
traditional RFC 4103, and at each pause in transmission five extra
packets with only redundant data will be sent compared to two extra
packets for the traditional RFC 4103 case.
4.1.1.8. RTP Mixer with RFC 5109 FEC and RFC 2198 redundancy and
separate sequence number in the packets
This method allows primary data from one source and redundant text
from other sources in each packet. The packet payload contains
primary data in a new "text/t140e" format, and redundant data in RFC
5109 FEC [RFC5109] format called "text/ulpfec". That means that the
redundant data contains the sequence number and the CSRC and other
characteristics from the RTP header when the data was sent as
primary. The redundancy can be sent at a selected number of packets
after when it was sent as primary, in order to improve the protection
against bursty packet loss. The redundancy level is recommended to
be the same as in original RFC 4103. The "text/t140e" format
contains a source-specific sequence number and the t140block.
RFC 4103 says that the protection against loss can be made by other
methods than plain redundancy, so this method is in line with that
statement.
Transmission is done as soon as there is new text available, but not
with shorter interval than 100 ms and not longer than 300 ms while
there is anything to send (new or redundant text).
When more consecutive packet loss than the number of generations of
redundant data appears, it is possible to deduce which sources lost
data when new data arrives from the sources. This is done by
monitoring the received source specific sequence numbers preceding
the text.
This is an example of how can indicate the format as "text/red" with
"text/t140e" as primary and "text/ulpfec" redundant data, with
traditional RFC 4103 with "text/red" with "text/t140" as redundant
data as a fallback.
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m=text 49170 RTP/AVP 98 101 100 102 103
a=rtpmap:98 red/1000
a=fmtp:98 100/102/102
a=rtpmap:102 ulpfec/1000
a=rtpmap:103 t140/1000
a=rtpmap:100 t140e/1000
a=rtpmap:101 red/1000
a=fmtp:101 103/103/103
a=fmtp:100 cps=200
The "text/ulpfec" format includes an indication of how far back the
redundancy belongs, making it possible to cover bursty packet loss
better than the other formats with short transmission intervals. For
real-time text, it is recommended to send three packets between the
primary and the redundant transmissions of text. That makes the
transmission cover between 500 and 1500 ms of bursty packet loss.
The variation is because of the varying packet interval between many
and one simultaneously transmitting source.
The "text/ulpfec" format has a number of parameters. One is the
length of the data to be protected which in this case must be the
whole t140block.
Pros:
The source switching performance is good. Text from 5 participants
can be transmitted within 500 ms.
Good recovery from bursty packet loss.
The method is based on an existing standard for FEC.
When more consecutive packet loss than the number of generations of
redundant data appears, it is possible to deduce the source of the
lost data when new text arrives from the source.
Cons:
Even if the switching performance is good, it is not as good as for
the method called "RTP Mixer with multiple primary data in each
packet" Section 4.1.1.6. With more than 5 simultaneously sending
sources, there will be a noticeable delay of text of over 500 ms,
with 100 ms added per simultaneous source. This is however beyond
the requirements and would be a concern only in congestion
situations.
The recovery procedure is a bit complex [RFC5109].
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There is more overhead in terms of extra data and extra packets sent
than in the other methods. With the recommended two redundant
generations of data, each packet will be 40 bytes longer than with
traditional RFC 4103, and at each pause in transmission five extra
packets with only redundant data will be sent compared to two extra
packets for the traditional RFC 4103 case.
A new text media subtype "text/t140e" needs to be registered.
The processing time in standard organisation will be long.
4.1.1.9. RTP Mixer indicating participants by a control code in the
stream
Text from all participants except the receiving one is transmitted
from the media mixer in the same RTP session and stream, thus all
using the same destination address/port combination, the same RTP
SSRC and , one sequence number series as described in Section 7.1 and
7.3 of RTP RFC 3550 [RFC3550] about the Mixer function. The sources
of the text in each RTP packet are identified by a new defined T.140
control code "c" followed by a unique identification of the source in
UTF-8 string format.
The receiver can use the string for presenting the source of text.
This method is on the RTP level described in RFC 7667, section 3.6.1
Media mixing mixer [RFC7667].
The inline coding of the source of text is applied in the data stream
itself, and an RTP mixer function is used for coordinating the
sources of text into one RTP stream.
Information uniquely identifying each user in the multi-party session
is placed as the parameter value "n" in the T.140 application
protocol function with the function code "c". The identifier shall
thus be formatted like this: SOS c n ST, where SOS and ST are coded
as specified in ITU-T T.140 [T140]. The "c" is the letter "c". The
n parameter value is a string uniquely identifying the source. This
parameter shall be kept short so that it can be repeated in the
transmission without concerns for network load.
A receiving endpoint is supposed to separate text items from the
different sources and identify and display them accordingly.
The conference server need to be allowed to decrypt/encrypt the
packet payload in order to check the source and repack the text.
Pros:
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If loss of packets occur, it is possible to recover text from
redundancy at loss of up to the number of redundancy levels carried
in the RFC 4103 [RFC4103]stream. (normally primary and two redundant
levels.
This method can be implemented with most RTP implementations.
The method can also be used with other transports than RTP
Cons:
The method implies a moderate load by the need to insert the source
often in the stream.
If more consecutive packet loss than the number of generations of
redundant data appears, it is not possible to deduce the source of
the totally lost data.
The mixer needs to be able to generate suitable and unique source
identifications which are suitable as labels for the sources.
Requires an extension on the ITU-T T.140 standard, best made by the
ITU.
There is a risk that the control code indicating the change of source
is lost and the result is false source indication of text.
The conference server need to be allowed to decrypt/encrypt the
packet payload.
4.1.1.10. Mixing for multi-party unaware user agents
Multi-party real-time text contents can be transmitted to multi-party
unaware user agents if source labelling and formatting of the text is
performed by a mixer. This method has the limitations that the
layout of the presentation and the format of source identification is
purely controlled by the mixer, and that only one source at a time is
allowed to present in real-time. Other sources need to be stored
temporarily waiting for an appropriate moment to switch the source of
transmitted text. The mixer controls the switching of sources and
inserts a source identifier in text format at the beginning of text
after switch of source. The logic of the mixer to detect when a
switch is appropriate should detect a number of places in text where
a switch can be allowed, including new line, end of sentence, end of
phrase, a period of inactivity, and a word separator after a long
time of active transmission.
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This method MAY be used when no support for multi-party awareness is
detected in the receiving endpoint.The base for his method is
described in RFC 7667, section 3.6.1 Media mixing mixer [RFC7667].
See [I-D.ietf-avtcore-multi-party-rtt-mix] for a procedure for mixing
RTT for a conference-unaware endpoint.
Pros:
Can be transmitted to conference-unaware endpoints.
Can be used with other transports than RTP
Cons:
Does not allow full real-time presentation of more than one source at
a time. Text from other sources will be delayed.
The only realistic presentation format is a style with the text from
the different sources presented with a text label indicating source,
and the text collected in a chat style presentation but with more
frequent turn-taking.
Endpoints often have their own system for adding labels to the RTT
presentation. In that case there will be two levels of labels in the
presentation, one for the mixer and one for the sources.
If loss of more packets than can be recovered by the redundancy
appears, it is not possible to detect which source was struck by the
loss. It is also possible that a source switch occurred during the
loss, and therefore a false indication of the source of text can be
provided to the user after such loss.
Because of all these cons, this method is not recommended be used as
the main method, but only as fallback and the last resort for
backwards interoperability with multi-party unaware endpoints.
The conference server need to be allowed to decrypt/encrypt the
packet payload.
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4.1.2. RTP-based bridging with minor RTT media contents reformatting by
the bridge
It may be desirable to send text in a multi-party setting in a way
that allows the text stream contents to be distributed without being
dealt with in detail in any central server. This approach may enable
end-to-end encryption. A number of such methods are described.
However, when writing this specification, no one of these methods
have a specified way of establishing the session by sdp.
4.1.2.1. One RTP stream for RTT per participant from the mixer
Within the RTP session, text from each participant is transmitted
from the RTP media bridge in a separate RTP stream, thus using the
same destination address/port combination, the same payload type
number (PT) but separate RTP SSRC parameters and sequence number
series as described in Section 7.1 and 7.2 of RTP RFC 3550 [RFC3550]
about the Translator function. The source of the text in each RTP
packet is identified by the SSRC parameter in the RTP packets,
containing the SSRC of the initial source of text.
A receiving and presenting endpoint is supposed to separate text
items from the different sources and identify and display them in a
suitable way.
This method is described in RFC 7667, section 3.5.1 Relay-transport
translator or 3.5.2 Media translator [RFC7667].
The identification of the source is made through the SSRC. The
translation to a readable label can be done by mapping to information
from the RTCP SDES CNAME and NAME packets as described in
RTP[RFC3550], and also through information in the text media member
in the conference notification described in RFC 4575 [RFC4575].
The sdp exchange for establishing this mixing type can be equal to
what is used for basic two-party use of RFC 4103 with just an added
attribute for indicating multi-party capability.
m=text 49170 RTP/AVP 98 103
a=rtpmap:98 red/1000
a=fmtp:98 103/103/103
a=rtpmap:103 t140/1000
a=fmtp:103 cps=150
a=RTT-mixing:RTP-translator
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A similar answer including the same RTT-mixing attribute would
indicate that multi-party coding can begin. An answer without the
same RTT-mixing attribute could result in diversion to use of the
mixing method for multi-party unaware endpoints Section 4.1.1.10 if
more than two parties are involved in the session.
The bridge can add new sources in the communication to a participant
by first sending a conference notification according to RFC 4575
[RFC4575] with the SSRC of the new source included in the
corresponding "text" media member, or by sending an RTCP message with
the new SSRC in an SDES packet.
A receiver should be prepared to receive such indications of new
streams being added to the multi-party session, so that the new SSRC
is not taken for a change in SSRC value for an already established
RTP stream.
Transmission, reception, packet loss recovery and text loss
indication is performed per source in the separate RTP streams in the
same way as in two-party sessions with RFC 4103 [RFC4575].
Text is recommended to be sent by the bridge as soon as it is
available for transmission, but not less than 250 ms after a previous
transmission. This will in many cases result in close to 0 added
delay by the bridge, because most RTT senders use a 300 ms
transmission interval.
It is sometimes said that this configuration is not supported by
current media declarations in sdp. RFC 3264 [RFC3264]specifies in
some places that one media description is supposed to describe just
one RTP media stream. However this is not directly referencing an
RTP stream, and use of multiple RTP streams in the same RTP session
is recommended in many other RFCs.
This confusion is clarified in RFC 5576 [RFC5576] section 3 by the
following statements:
"The term "media stream" does not appear in the SDP specification
itself, but is used by a number of SDP extensions, for instance,
Interactive Connectivity Establishment (ICE) [ICE], to denote the
object described by an SDP media description. This term is
unfortunately rather confusing, as the RTP specification [RFC3550]
uses the term "media stream" to refer to an individual media source
or RTP packet stream, identified by an SSRC, whereas an SDP media
stream describes an entire RTP session, which can contain any number
of RTP sources."
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In most cases, it will be sufficient that new sources are introduced
with a conference notification or RTCP message. However, RFC 5576
[RFC5576] specifies attributes which may be used to more explicitly
announce new sources or restart of earlier established RTP streams.
This method is encouraged by draft-ietf-avtcore-multiplex-guidelines
[I-D.ietf-avtcore-multiplex-guidelines] section 5.2.
One way of operation will be that the bridge receives text packets
from the source and handles any text recovery and indication of loss
needed before queueing the resulting clean text for transmission from
the bridge to the receivers. However, that method requires the mixer
to decrypt the payload of the packets and makes end-to-end encryption
impossible.
It may however also be possible for the bridge to just convey the
packet contents as received from the sources, with minor adjustments
in the RTP header, and let the receiving endpoint handle all aspects
of recovery and indication of loss, even for the source to bridge
path. In that case also the sequence number sequence must be
maintained as it was at reception in the bridge at least regarding
gaps in the sequence. This mode needs further study before
application.
Pros:
This method may be designed so that end-to-end encryption is enabled.
This method is a natural way to do multi-party bridging with RFC 4103
based RTT.
This method has moderate overhead in terms of work for the mixer, but
high in terms of packet transmission rate. Five sources sending
simultaneously cause the bridge to send 15 packets per second to each
receiver.
When loss of packets occur, it is possible to recover text from
redundancy at loss of up to the number of redundancy levels carried
in the RFC 4103 [RFC4103] stream(normally primary and two redundant
levels).
More loss than what can be recovered, can be detected and the marker
for text loss can be inserted in the correct stream.
It may be possible in some scenarios to keep the text encrypted
through the Translator.
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Minimal delay. The delay can often be kept close to 0 with at least
5 simultaneous sending participants.
Cons:
There are RTP implementations not supporting the Translator model.
They will need to use the fall-back to multi-party-unaware mixing or
another method based on RTP-mixer. An investigation about how common
this lack of support is is needed before the method is used.
The processing time in standard organisation will be long.
With many simultaneous sending sources, the total rate of packets
will be high, and can cause congestion. The requirement to handle 3
simultaneous sources in this specification will cause 10 packets per
second that is manageable in most cases, e.g. considering that audio
usually use 50 packets per second.
4.1.2.2. Selective Forwarding Middlebox
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 is described in the
previous section. Another 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. The idea of SFM
is that the mixer selects a limited number of sources to be conveyed
to the participants while other media streams are discarded. This
causes very good efficiency for the audio and video media which are
transmitted continuously from the sources.
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 all
normal conditions. It needs however be able to vary which sources
are conveyed depending on which users are active transmitting at the
moment.
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
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act per stream. The relation between the SSRC and the source would
need to be conveyed in some specified way, e.g. in the CSRC.
Recovery and loss detection would preferably be based on sequence
number gap detection. Thus sequence number gaps in the incoming
stream to the mixer would need to be reflected in the stream to the
participant and no new gaps created by the mixer, even if the
sequence number series may be different.
Pros:
This method may be designed so that end-to-end encryption is enabled.
This method is a natural way to do multi-party bridging with RFC 4103
based RTT.
This method has moderate overhead in terms of work for the mixer, but
high in terms of packet transmission rate. Five sources sending
simultaneously cause the bridge to send 15 packets per second to each
receiver.
When loss of packets occur, it is possible to recover text from
redundancy at loss of up to the number of redundancy levels carried
in the RFC 4103 [RFC4103] stream(normally primary and two redundant
levels).
More loss than what can be recovered, can be detected and the marker
for text loss can be inserted in the correct stream.
Minimal delay. The delay can often be kept close to 0 with at least
5 simultaneous sending participants.
Cons:
There are RTP implementations not supporting the SFM method. They
will need to use the fall-back to multi-party-unaware mixing or
another method based on RTP-mixer.
With very rarely occurring high number of simultaneous sending
sources, the SFM will need to discard text from some sources in order
to keep the total rate of packets at a suitable level. That can
cause confusion.
This method requires a lot of further specification.
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4.1.2.3. Distributing packets in an end-to-end encryption structure
In order to achieve end-to-end encryption, it is possible to let the
packets from the sources just pass though a central distributor, and
handle the security agreements between the participants.
Specifications exist for a framework with this functionality for
application on RTP based conferences in
[I-D.ietf-perc-private-media-framework]. The RTP flow and mixing
characteristics has similarities with the method described under "RTP
Translator sending one RTT stream per participant" above. RFC 4103
RTP streams [RFC4103] would fit into the structure and it would
provide a base for end-to-end encrypted rtt multi-party conferencing.
Pros:
Good security
Straightforward multi-party handling.
Cons:
Does not operate under the usual SIP central conferencing
architecture.
Requires the participants to perform a lot of key handling.
Is work in progress when this is written.
4.1.2.4. Mesh of RTP endpoints
Text from all participants are transmitted directly to all others in
one RTP session, without a central bridge. The sources of the text
in each RTP packet are identified by the source network address and
the SSRC.
This method is described in RFC 7667, section 3.4 Point to multi-
point using mesh [RFC7667].
Pros:
When loss of packets occur, it is possible to recover text from
redundancy at loss of up to the number of redundancy levels carried
in the RFC 4103 [RFC4103] stream. (normally primary and two redundant
levels.
This method can be implemented with most RTP implementations.
Transmitted text can also be used with other transports than RTP
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Cons:
This model is not described in IMS, NENA and EENA specifications, and
does therefore not meet the requirements.
Requires a drastically increasing number of connections when the
number of participants increase.
4.1.2.5. Multiple RTP sessions, one for each participant
Text from all participants are transmitted directly to all others in
one RTP session each, without a central bridge. Each session is
established with a separate media description in SDP. The sources of
the text in each RTP packet are identified by the source network
address and the SSRC.
Pros:
When loss of packets occur, it is possible to recover text from
redundancy at loss of up to the number of redundancy levels carried
in the RFC 4103 [RFC4103] stream. (normally primary and two redundant
levels.
Complete loss of text can be indicated in the received stream.
This method can be implemented with most RTP implementations.
End-to-end encryption is achievable.
Cons:
This method is not described in IMS, NENA and ETSI specifications and
does therefore not meet the requirements.
A lot of network resources are spent on setting up separate sessions
for each participant.
5. Preferred RTP-based multi-party RTT transport method
For RTP transport of RTT using RTP-mixer technology, one method for
multi-party mixing and transport stand out as fulfilling the goals
best and is therefore recommended. That is: "RTP Mixer interleaving
packets, receiver using timestamp to recover from loss"
Section 4.1.1.4
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For RTP transport in separate streams or sessions, no current
recommendation can be made. A bridging method in the process of
standardisation with interesting characteristics is the end-to-end
encryption model "perc" Section 4.1.2.3.
6. Session control of RTP-based multi-party RTT sessions
General session control aspects for multi-party sessions are
described in RFC 4575 [RFC4575] A Session Initiation Protocol (SIP)
Event Package for Conference State, and RFC 4579 [RFC4579] Session
Initiation Protocol (SIP) Call Control - Conferencing for User
Agents. The nomenclature of these specifications are used here.
The procedures for a multi-party aware model for RTT-transmission
shall only be applied if a capability exchange for multi-party aware
real-time text transmission has been completed and a supported method
for multi-party real-time text transmission can be negotiated.
A method for detection of conference-awareness for centralized SIP
conferencing in general is specified in RFC 4579 [RFC4579]. The
focus sends the "isfocus" feature tag in a SIP Contact header. This
causes the conference-aware endpoint to subscribe to conference
notifications from the focus. The focus then sends notifications to
the endpoint about entering and disappearing conference participants
and their media capabilities. The information is carried XML-
formatted in a 'conference-info' block in the notification according
to RFC 4575 [RFC4575]. The mechanism is described in detail in RFC
4575 [RFC4575].
Before a conference media server starts sending multi-party RTT to an
endpoint, a verification of its ability to handle multi-party RTT
must be made. A decision on which mechanism to use for identifying
text from the different participants must also be taken, implicitly
or explicitly. These verifications and decisions can be done in a
number of ways. The most apparent ways are specified here and their
pros and cons described. One of the methods is selected to be the
one to be used by implementations of the centralized conference model
according to this specification.
6.1. Implicit RTT multi-party capability indication
Capability for RTT multi-party handling can be decided to be
implicitly indicated by session control items.
The focus may implicitly indicate muti-party RTT capability by
including the media child with value "text" in the RFC 4575 [RFC4575]
conference-info provided in conference notifications.
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An endpoint may implicitly indicate multi-party RTT capability by
including the text media in the SDP in the session control
transactions with the conference focus after the subscription to the
conference has taken place.
The implicit RTT capability indication means for the focus that it
can handle multi-party RTT according to the preferred method
indicated in the RTT multi-party methods section above.
The implicit RTT capability indication means for the endpoint that it
can handle multi-party RTT according to the preferred method
indicated in the RTT multi-party methods section above.
If the focus detects that an endpoint implicitly declared RTT multi-
party capability, it SHALL provide RTT according to the preferred
method.
If the focus detects that the endpoint does not indicate any RTT
multi-party capability, then it shall either provide RTT multi-party
text in the way specified for conference-unaware endpoint above, or
refuse to set up the session.
If the endpoint detects that the focus has implicitly declared RTT
multi-party capability, it shall be prepared to present RTT in a
multi-party fashion according to the preferred method.
Pros:
Acceptance of implicit multi-party capability implies that no
standardisation of explicit RTT multi-party capability exchange is
required.
Cons:
If other methods for multi-party RTT are to be used in the same
implementation environment as the preferred ones, then capability
exchange needs to be defined for them.
Cannot be used outside a strictly applied SIP central conference
model.
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6.2. RTT multi-party capability declared by SIP media-tags
Specifications for RTT multi-party capability declarations can be
agreed for use as SIP media feature tags, to be exchanged during SIP
call control operation according to the mechanisms in RFC 3840
[RFC3840] and RFC 3841 [RFC3841]. Capability for the RTT Multi-party
capability is then indicated by the media feature tag "rtt-mix", with
a set of possible values for the different possible methods.
The possible values in the list may for example be:
rtp-mixer
perc
rtp-mixer indicates capability for using the RTP-mixer based
presentation of multi-party text.
perc indicates capability for using the perc based transmission of
multi-party text.
Example: Contact: <sip:a2@beco.example.com>
;methods="INVITE,ACK,OPTIONS,BYE,CANCEL"
;+sip.rtt-mix="rtp-mixer"
If, after evaluation of the alternatives in this specification, only
one mixing method is selected to be brought to implementation, then
the media tag can be reduced to a single tag with no list of values.
An offer-answer exchange should take place and the common method
selected by the answering party shall be used in the session with
that UA.
When no common method is declared, then only the fallback method for
multi-party unaware participants can be used, or the session dropped.
If more than one text media section is included in SDP, all must be
capable of using the declared RTT multi-party method.
Pros:
Provides a clear decision method.
Can be extended with new mixing methods.
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Can guide call routing to a suitable capable focus.
Cons:
Requires standardization and IANA registration.
Is not stream specific. If more than one text stream is specified,
all must have the same type of multi-party capability.
Cannot be used in the WebRTC environment.
6.3. SDP media attribute for RTT multi-party capability indication
An attribute can be specified on media level, to be used in text
media SDP declarations for negotiating RTT multi-party capabilities.
The attribute can have the name "rtt-mixing".
More than one attribute can be included in one media description.
The attribute can have a value. The value can for example be:
rtp-mixer
rtp-translator
perc
rtp-mixer indicates capability for using the RTP-mixer and CSRC-list
based mixing of multi-party text.
rtp-translator indicates capability for using the RTP-translator
based mixing
perc indicates capability for using the perc based transmission of
multi-party text.
An offer-answer exchange should take place and the common method
selected by the answering party shall be used in the session with
that endpoint.
When no common method is declared, then only the fallback method for
multi-party unaware endpoints can be used.
Example: a=rtt-mixing:rtp-mixer
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If, after evaluation of the alternatives in this specification, only
one mixing method is selected to be brought to implementation, then
the attribute can be reduced to a single attribute with no list of
values.
Pros:
Provides a clear decision method.
Can be extended with new mixing methods.
Can be used on specific text media.
Can be used also for SDP-controlled WebRTC sessions with multiple
streams in the same data channel.
Cons:
Requires standardization and IANA registration.
Cannot guide SIP routing.
6.4. Simplified SDP media attribute for RTT multi-party capability
indication
An attribute can be specified on media level, to be used in text
media SDP declarations for negotiating RTT multi-party capabilities.
The attribute can have a name suitable for the selected method and no
value. It would be selected and used if only one method for multi-
party rtt is brought forward from this specification, and the other
left unspecified for now or found to be possible to negotiate in
another way.
An offer-answer exchange should take place and if both parties
specify rtt-mixing capability with the same attribute, the selected
mixing method shall be used.
When no common method is declared, then only the fallback method for
multi-party unaware endpoints can be used, or the session not
accepted for multi-party use.
Example: a=rtt-mix
Pros:
Provides a clear decision method.
Very simple syntax and semantics.
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Can be used on specific text media.
Cons:
Requires standardization and IANA registration.
If another RTT mixing method is also specified in the future, then
that method may also need to specify and register its own attribute,
instead of if an attribute with a parameter value is used, when only
an addition of a new possible value is needed.
Cannot guide SIP routing.
6.5. SDP format parameter for RTT multi-party capability indication
An FMTP format parameter can be specified for the RFC 4103
[RFC4103]media, to be used in text media SDP declarations for
negotiating RTT multi-party capabilities. The parameter can have the
name "rtt-mixing", with one or more of its possible values.
The possible values in the list are:
rtp-mixer
perc
rtp-mixer indicates capability for using the RTP-mixer based mixing
and presentation of multi-party text using the CSRC-list.
perc indicates capability for using the perc based transmission of
multi-party text.
Example: a=fmtp 96 98/98/98 rtt-mixing=rtp-mixer
If, after evaluation of the alternatives in this specification, only
one mixing method is selected to be brought to implementation, then
the parameter can be reduced to a single parameter with no list of
values.
An offer-answer exchange should take place and the common method
selected by the answering party shall be used in the session with
that UA.
When no common method is declared, then only the fallback method can
be used, or the session denied.
Pros:
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Provides a clear decision method.
Can be extended with new mixing methods.
Can be used on specific text media.
Can be used also for SDP-controlled WebRTC sessions with multiple
streams in the same data channel.
Cons:
Requires standardization and IANA registration.
May cause interop problems with current RFC4103 [RFC4103]
implementations not expecting a new fmtp-parameter.
Cannot guide SIP routing.
6.6. A text media subtype for support of multi-party rtt
Indicating a specific text media subtype in SDP is a straightforward
way for negotiating multi-party capability. Especially if there are
format differences from the "text/red" and "text/t140" formats of
RFC4103 [RFC4103], then this is a natural way to do the negotiation
for multi-party rtt.
Pros:
No extra efforts if a new format is needed anyway.
Cons:
None specific to using the format indication for negotiation of
multi-party capability. But only feasible if a new format is needed
anyway.
6.7. Preferred capability declaration method for RTP-based transport.
If the preferred transport method is one with a specific media
subtype in sdp, then specification by media subtype is preferred.
If this would not be the case, then the preferred capability
declaration method would be the one with a specific SDP attribute for
the selected mixing method Section 6.4 because it is straightforward.
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6.8. Identification of the source of text for RTP-based solutions
The main way to identify the source of text in the RTP based solution
is by the SSRC of the sending participant. In the RTP-mixer
solution, this SSRC is included in the CSRC list of the transmitted
packets. Further identification that may be needed for better
labelling of received text may be achieved from a number of sources.
It may be the RTCP SDES CNAME and NAME reports, and in the conference
notification data (RFC 4575) [RFC4575].
As soon as a new member is added to the RTP session, its
characteristics should be transmitted in RTCP SDES CNAME and NAME
reports according to section 6.5 in RFC 3550 [RFC3550]. The
information about the participant should also be included in the
conference data including the text media member in a notification
according to RFC 4575 [RFC4575].
The RTCP SDES report, SHOULD contain identification of the source
represented by the SSRC/CSRC identifier. This identification MUST
contain the CNAME field and MAY contain the NAME field and other
defined fields of the SDES report.
A focus UA SHOULD primarily convey SDES information received from the
sources of the session members. When such information is not
available, the focus UA SHOULD compose SSRC/CSRC, CNAME and NAME
information from available information from the SIP session with the
participant.
Provision of detailed information in the NAME field has security
implications, especially if provided without encryption.
7. RTT bridging in WebRTC
Within WebRTC, real-time text is specified to be carried in WebRTC
data channels as specified in
[I-D.ietf-mmusic-t140-usage-data-channel]. A few ways to handle
multi-party RTT are mentioned briefly. They are repeated below.
7.1. RTT bridging in WebRTC with one data channel per source
A straightforward way to handle multi-party RTT is for the bridge to
open one T.140 data channel per source towards the receiving
participants.
The stream-id forms a unique stream identification.
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The identification of the source is made through the Label property
of the channel, and session information belonging to the source. The
endpoint can compose a readable label for the presentation from this
information.
Pros:
This is a straightforward solution.
The load per source is low.
Cons:
With a high number of participants, the overhead of establishing and
maintaining the high number of data channels required may be high,
even if the load per channel is low.
7.2. RTT bridging in WebRTC with one common data channel
A way to handle multi-party RTT in WebRTC is for the bridge combine
text from all sources into one data channel and insert the sources in
the stream by a T.140 control code for source.
This method is described in a corresponding section for RTP
transmission above in Section 4.1.1.9.
The identification of the source is made through insertion in the
beginning of each text transmission from a source of a control code
extension "c" followed by a string representing the source, framed by
the control code start and end flags SOS and ST (See ITU-T T.140
[T140]).
A receiving endpoint is supposed to separate text items from the
different sources and identify and display them in a suitable way.
The endpoint does not always display the source identification in the
received text at the place where it is received, but has the
information as a guide for planning the presentation of received
text. A label corresponding to the source identification is
presented when needed depending on the selected presentation style.
Pros:
This solution has relatively low overhead on session and network
level
Cons:
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This solution has higher overhead on the media contents level than
the WebRTC solution above.
Standardisation of the new control code "c" in ITU-T T.140 [T140] is
required.
The conference server need to be allowed to decrypt/encrypt the data
channel contents.
7.3. Preferred rtt multi-party method for WebRTC
For WebRTC, one method is to prefer because of the simplicity. So,
for WebRTC, the method to implement for multi-party RTT with multi-
party aware parties when no other method is explicitly agreed between
implementing parties is: "RTT bridging in WebRTC with one data
channel per source" Section 7.1.
8. Presentation of multi-party text
All session participants with RTP based transport MUST observe the
SSRC/CSRC field of incoming text RTP packets, and make note of which
source they came from in order to be able to present text in a way
that makes it easy to read text from each participant in a session,
and get information about the source of the text.
In the WebRTC case, the Label parameter and other provided endpoint
information should be used for the same purpose.
8.1. Associating identities with text streams
A source identity SHOULD be composed from available information
sources and displayed together with the text as indicated in ITU-T
T.140 Appendix[T140].
The source identity should primarily be the NAME field from incoming
SDES packets. If this information is not available, and the session
is a two-party session, then the T.140 source identity SHOULD be
composed from the SIP session participant information. For multi-
party sessions the source identity may be composed by local
information if sufficient information is not available in the
session.
Applications may abbreviate the presented source identity to a
suitable form for the available display.
Applications may also replace received source information with
internally used nicknames.
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8.2. Presentation details for multi-party aware endpoints.
The multi-party aware endpoint should after any action for recovery
of data from lost packets, separate the incoming streams and present
them according to the style that the receiving application supports
and the user has selected. The decisions taken for presentation of
the multi-party interchange shall be purely on the receiving side.
The sending application must not insert any item in the stream to
influence presentation that is not requested by the sending
participant.
8.2.1. Bubble style presentation
One often used style is to present real-time text in chunks in
readable bubbles identified by labels containing names of sources.
Bubbles are placed in one column in the presentation area and are
closed and moved upwards in the presentation area after certain items
or events, when there is also newer text from another source that
would go into a new bubble. The text items that allows bubble
closing are any character closing a phrase or sentence followed by a
space or a timeout of a suitable time (about 10 seconds).
Real-time active text sent from the local user should be presented in
a separate area. When there is a reason to close a bubble from the
local user, the bubble should be placed above all real-time active
bubbles, so that the time order that real-time text entries were
completed is visible.
Scrolling is usually provided for viewing of recent or older text.
When scrolling is done to an earlier point in the text, the
presentation shall not move the scroll position by new received text.
It must be the decision of the local user to return to automatic
viewing of latest text actions. It may be useful with an indication
that there is new text to read after scrolling to an earlier position
has been activated.
The presentation area may become too small to present all text in all
real-time active bubbles. Various techniques can be applied to
provide a good overview and good reading opportunity even in such
situations. The active real-time bubble may have a limited number of
lines and if their contents need more lines, then a scrolling
opportunity within the real-time active bubble is provided. Another
method can be to only show the label and the last line of the active
real-time bubble contents, and make it possible to expand or compress
the bubble presentation between full view and one line view.
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Erasures require special consideration. Erasure within a real-time
active bubble is straightforward. But if erasure from one
participant affects the last character before a bubble, the whole
previous bubble becomes the actual bubble for real-time action by
that participant and is placed below all other bubbles in the
presentation area. If the border between bubbles was caused by the
CRLF characters (instead of the normal "Line Separator"), only one
erasure action is required to erase this bubble border. When a
bubble is closed, it is moved up, above all real-time active bubbles.
A three-party view is shown in this example .
_________________________________________________
| |^|
| |-|
|[Alice] Hi, Alice here. | |
| | |
|[Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris. | |
| I thought you should be here. | |
| | |
|[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|[Bob] And I on Wednesday evening. | |
| | |
|[Alice] Can we meet on Thursday evening? | |
| | |
|[Eve] Yes, definitely. How about 7pm. | |
| at the entrance of the restaurant | |
| Le Lion Blanc? | |
|[Eve] we can have dinner and then take a walk | |
| | |
| <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 1: Three-party call with bubble style.
Figure 1: Example of a three-party call presented in the bubble
style.
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8.2.2. Other presentation styles
Other presentation styles than the bubble style may be arranged and
appreciated by the users. In a video conference one way may be to
have a real-time text area below the video view of each participant.
Another view may be to provide one column in a presentation area for
each participant and place the text entries in a relative vertical
position corresponding to when text entry in them was completed. The
labels can then be placed in the column header. The considerations
for ending and moving and erasure of entered text discussed above for
the bubble style are valid also for these styles.
This figure shows how a coordinated column view MAY be presented.
_____________________________________________________________________
| Bob | Eve | Alice |
|____________________|______________________|_______________________|
| | |I will arrive by TGV. |
|My flight is to Orly| |Convenient to the main |
| |Hi all, can we plan |station. |
| |for the seminar? | |
|Eve, will you do | | |
|your presentation on| | |
|Friday? |Yes, Friday at 10. | |
|Fine, wo | |We need to meet befo |
|___________________________________________________________________|
Figure 2: A coordinated column-view of a three-party session with
entries ordered in approximate time-order.
9. Presentation details for multi-party unaware endpoints.
Multi-party unaware endpoints are prepared only for presentation of
two sources of text, the local user and a remote user. If mixing for
multi-party unaware endpoints is to be supported, in order to enable
some multi-party communication with such endpoint, the mixer need to
plan the presentation and insert labels and line breaks before
lables. Many limitations appear for this presentation mode, and it
must be seen as a fallback and a last resort.
A procedure for presenting RTT to a conference-unaware endpoint is
included in [I-D.ietf-avtcore-multi-party-rtt-mix]
10. Security Considerations
The security considerations valid for RFC 4103 [RFC4103] and RFC 3550
[RFC3550] are valid also for the multi-party sessions with text.
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11. IANA Considerations
The items for indication and negotiation of capability for multi-
party rtt should be registered with IANA in the specifications where
they are specified in detail.
12. Congestion considerations
The congestion considerations described in RFC 4103 [RFC4103] are
valid also for the recommended RTP-based multi-party use of the real-
time text transport. A risk for congestion may appear if a number of
conference participants are active transmitting text simultaneously,
because the recommended RTP-based multi-party transmission method
does not allow multiple sources of text to contribute to the same
packet.
In situations of risk for congestion, the Focus UA MAY combine
packets from the same source to increase the transmission interval
per source up to one second. Local conference policy in the Focus UA
may be used to decide which streams shall be selected for such
transmission frequency reduction.
13. Acknowledgements
Arnoud van Wijk for contributions to an earlier, expired draft of
this memo.
14. Change history
14.1. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-06
Addition of the Selective Forwarding Middlebox SFM among the methods
with multiple RTP streams, to match the contents of draft-ietf-
avtcore-multi-party-rtt-mix-12.
14.2. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-05
Modify the solution changing source in every packet in the RTP-mixer
solution, and base recovery on analyzing timestamp and make it the
recommended one. Aligned with the recommendation in draft-ietf-
avtcore-multi-party-rtt-mix-10.
14.3. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-04
Change name of simplified sdp attribute to "rtt-mix" to match a
change in the draft draft-ietf-avtcore-multi-party-rtt-mix-09.
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14.4. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-03
Modified info on the method with RFC 4103 format and sdp attribute
"rtt-mix-rtp-mixer".
Increased the performance requirements section.
Inserted recommendations, with emphasis on ease of implementation and
ease of standardisation.
14.5. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-02
Added detail in the section on RTP translator model alternative
4.1.2.1.
14.6. Changes to draft-hellstrom-avtcore-multi-party-rtt-solutions-01
Added three more methods for RTP-mixer mixing. Two RFC 5109 FEC
based and another with modified data header to detect source of
completely lost text.
Separated RTP-based and WebRTC based solutions.
Deleted the multi-party-unaware mixing procedure appendix. It is now
included in the draft draft-ietf-avtcore-multi-party-rtt-mix. Kept a
section with a reference to the new place.
14.7. Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to draft-
hellstrom-avtcore-multi-party-rtt-solutions-00
Add discussion about switching performance, as discussed in avtcore
on March 13.
Added that a decrease of transmission interval to 100 ms increases
switching performance by a factor 3, but still not sufficient.
Added that the CSRC-list method also uses 100 milliseconds
transmission interval.
Added the method with multiple primary text in each packet.
Added the timestamp-based method for rtp-mixing proposed by James
Hamlin on March 14.
Corrected the chat style presentation example picture. Delete a few
"[mix]".
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14.8. Changes from version draft-hellstrom-mmusic-multi-party-rtt-01 to
-02
Change from a general overview to overview with clear
recommendations.
Splits text coordination methods in three groups.
Recommends rtt-mixer with sources in CSRC-list but refers to its spec
for details.
Shortened Appendix with conference-unaware example.
Cleaned up preferences.
Inserted pictures of screen-views.
15. References
15.1. 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>.
15.2. Informative References
[EN301549] ETSI, "EN 301 549. Accessibility requirements for ICT
products and services", November 2019,
<https://www.etsi.org/deliver/
etsi_en/301500_301599/301549/03.01.01_60/
en_301549v030101p.pdf>.
[I-D.ietf-avtcore-multi-party-rtt-mix]
Hellstrom, G., "RTP-mixer formatting of multi-party Real-
time text", Work in Progress, Internet-Draft, draft-ietf-
avtcore-multi-party-rtt-mix-10, 18 November 2020,
<https://tools.ietf.org/html/draft-ietf-avtcore-multi-
party-rtt-mix-10>.
[I-D.ietf-avtcore-multiplex-guidelines]
Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
and R. Even, "Guidelines for using the Multiplexing
Features of RTP to Support Multiple Media Streams", Work
in Progress, Internet-Draft, draft-ietf-avtcore-multiplex-
guidelines-12, 16 June 2020, <https://tools.ietf.org/html/
draft-ietf-avtcore-multiplex-guidelines-12>.
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[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>.
[I-D.ietf-perc-private-media-framework]
Jones, P., Benham, D., and C. Groves, "A Solution
Framework for Private Media in Privacy Enhanced RTP
Conferencing (PERC)", Work in Progress, Internet-Draft,
draft-ietf-perc-private-media-framework-12, 5 June 2019,
<https://tools.ietf.org/html/draft-ietf-perc-private-
media-framework-12>.
[NENAi3] NENA, "NENA-STA-010.2-2016. Detailed Functional and
Interface Standards for the NENA i3 Solution", October
2016, <https://www.nena.org/page/i3_Stage3>.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J.C., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
DOI 10.17487/RFC2198, September 1997,
<https://www.rfc-editor.org/info/rfc2198>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[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>.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
"Indicating User Agent Capabilities in the Session
Initiation Protocol (SIP)", RFC 3840,
DOI 10.17487/RFC3840, August 2004,
<https://www.rfc-editor.org/info/rfc3840>.
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[RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
Preferences for the Session Initiation Protocol (SIP)",
RFC 3841, DOI 10.17487/RFC3841, August 2004,
<https://www.rfc-editor.org/info/rfc3841>.
[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>.
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353,
DOI 10.17487/RFC4353, February 2006,
<https://www.rfc-editor.org/info/rfc4353>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <https://www.rfc-editor.org/info/rfc4575>.
[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents",
BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006,
<https://www.rfc-editor.org/info/rfc4579>.
[RFC4597] Even, R. and N. Ismail, "Conferencing Scenarios",
RFC 4597, DOI 10.17487/RFC4597, August 2006,
<https://www.rfc-editor.org/info/rfc4597>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <https://www.rfc-editor.org/info/rfc5109>.
[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>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<https://www.rfc-editor.org/info/rfc5576>.
[RFC6443] Rosen, B., Schulzrinne, H., Polk, J., and A. Newton,
"Framework for Emergency Calling Using Internet
Multimedia", RFC 6443, DOI 10.17487/RFC6443, December
2011, <https://www.rfc-editor.org/info/rfc6443>.
Hellstrom Expires 20 June 2021 [Page 46]
Internet-Draft Real-time text multi-party solutions December 2020
[RFC6881] Rosen, B. and J. Polk, "Best Current Practice for
Communications Services in Support of Emergency Calling",
BCP 181, RFC 6881, DOI 10.17487/RFC6881, March 2013,
<https://www.rfc-editor.org/info/rfc6881>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[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>.
[TS103479] ETSI, "TS 103 479. Emergency communications (EMTEL); Core
elements for network independent access to emergency
services", December 2019, <https://www.etsi.org/deliver/
etsi_ts/103400_103499/103479/01.01.01_60/
ts_103479v010101p.pdf>.
[TS22173] 3GPP, "IP Multimedia Core Network Subsystem (IMS)
Multimedia Telephony Service and supplementary services;
Stage 1", 3GPP TS 22.173 17.1.0, 20 December 2019,
<http://www.3gpp.org/ftp/Specs/html-info/22173.htm>.
[TS24147] 3GPP, "Conferencing using the IP Multimedia (IM) Core
Network (CN) subsystem; Stage 3", 3GPP TS 24.147 16.0.0,
19 December 2019,
<http://www.3gpp.org/ftp/Specs/html-info/24147.htm>.
Author's Address
Gunnar Hellstrom
Gunnar Hellstrom Accessible Communication
Esplanaden 30
SE-136 70 Vendelso
Sweden
Phone: +46 708 204 288
Email: gunnar.hellstrom@ghaccess.se
Hellstrom Expires 20 June 2021 [Page 47]