Internet Engineering Task Force G. Hellstrom
Internet-Draft Omnitor
Intended status: Best Current Practice March 28, 2020
Expires: September 29, 2020
Real-time text solutions for multi-party sessions
draft-hellstrom-avtcore-multi-party-rtt-solutions-00
Abstract
This document specifies methods for Real-Time Text (RTT) media
handling in multi-party calls. The main RTT transport is to carry
Real-Time text by the RTP protocol in a time-sampled mode according
to RFC 4103 [RFC4103] . 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.
Status of This Memo
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provisions of BCP 78 and BCP 79.
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working documents as Internet-Drafts. The list of current Internet-
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Requirements Language . . . . . . . . . . . . . . . . . . 4
2. Centralized conference model . . . . . . . . . . . . . . . . 5
3. Requirements on multi-party RTT . . . . . . . . . . . . . . . 5
4. Coordination of text RTP streams . . . . . . . . . . . . . . 7
4.1. RTP-based solutions with a central mixer . . . . . . . . 7
4.1.1. RTP Mixer using default RFC 4103 methods . . . . . . 7
4.1.2. RTP Mixer using the default method but decreased
transmission interval . . . . . . . . . . . . . . . . 8
4.1.3. RTP Mixer with frequent transmission and indicating
sources in CSRC-list . . . . . . . . . . . . . . . . 8
4.1.4. RTP Mixer using timestamp to identify redundancy . . 10
4.1.5. RTP Mixer with multiple primary data in each packet . 10
4.1.6. RTP Mixer indicating participants by a control code
in the stream . . . . . . . . . . . . . . . . . . . . 11
4.1.7. Mixing for multi-party unaware user agents . . . . . 12
4.2. RTP-based bridging with RTT media contents not touched by
the bridge . . . . . . . . . . . . . . . . . . . . . . . 14
4.2.1. RTP Translator sending one RTT stream per participant 14
4.2.2. Distributing packets in an end-to-end encryption
structure . . . . . . . . . . . . . . . . . . . . . . 15
4.2.3. Mesh of RTP endpoints . . . . . . . . . . . . . . . . 15
4.2.4. Multiple RTP sessions, one for each participant . . . 16
4.3. RTT bridging in WebRTC . . . . . . . . . . . . . . . . . 17
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4.3.1. RTT bridging in WebRTC with one data channel per
source . . . . . . . . . . . . . . . . . . . . . . . 17
4.3.2. RTT bridging in WebRTC with one common data channel . 17
5. Preferred multi-party RTT transport method . . . . . . . . . 18
6. Session control of multi-party RTT sessions . . . . . . . . . 18
6.1. Implicit RTT multi-party capability indication . . . . . 19
6.2. RTT multi-party capability declared by SIP media-tags . . 20
6.3. SDP media attribute for RTT multi-party capability
indication . . . . . . . . . . . . . . . . . . . . . . . 21
6.4. Simplified SDP media attribute for RTT multi-party
capability indication . . . . . . . . . . . . . . . . . . 23
6.5. SDP format parameter for RTT multi-party capability
indication . . . . . . . . . . . . . . . . . . . . . . . 23
6.6. Preferred capability declaration method. . . . . . . . . 25
7. Identification of the source of text . . . . . . . . . . . . 25
8. Presentation of multi-party text . . . . . . . . . . . . . . 25
8.1. Associating identities with text streams . . . . . . . . 26
8.2. Presentation details for multi-party aware endpoints. . . 26
8.2.1. Bubble style presentation . . . . . . . . . . . . . . 26
8.2.2. Other presentation styles . . . . . . . . . . . . . . 28
9. Presentation details for multi-party unaware endpoints. . . . 29
10. Security Considerations . . . . . . . . . . . . . . . . . . . 29
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 29
12. Congestion considerations . . . . . . . . . . . . . . . . . . 29
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 30
14. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
14.1. Changes from draft-hellstrom-mmusic-multi-party-rtt-02
to draft-avtcore-multi-party-rtt-solutions-00 . . . . . 30
14.2. Changes from version draft-hellstrom-mmusic-multi-party-
rtt-01 to -02 . . . . . . . . . . . . . . . . . . . . . 30
15. References . . . . . . . . . . . . . . . . . . . . . . . . . 31
15.1. Normative References . . . . . . . . . . . . . . . . . . 31
15.2. Informative References . . . . . . . . . . . . . . . . . 31
Appendix A. Mixing for a multi-party unaware endpoint . . . . . 33
A.1. Short description . . . . . . . . . . . . . . . . . . . . 34
A.2. Functionality goals and drawbacks . . . . . . . . . . . . 34
A.3. Definitions . . . . . . . . . . . . . . . . . . . . . . . 35
A.4. Presentation level procedures . . . . . . . . . . . . . . 37
A.4.1. Structure . . . . . . . . . . . . . . . . . . . . . . 37
A.4.2. Action on reception . . . . . . . . . . . . . . . . . 37
A.5. Display examples . . . . . . . . . . . . . . . . . . . . 41
A.6. References for this Appendix . . . . . . . . . . . . . . 42
A.7. Acknowledgement for the appendix . . . . . . . . . . . . 43
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 43
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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.
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
[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].
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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.
The recommended method for using RTT in a centralized conference
model is specified in [I-D.hellstrom-avtcore-multi-party-rtt-source]
based on the recommendations in the present 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
The following 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]).
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The transmission interval for text must 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 shall
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]
R7: 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 RFC
5194.[RFC5194]
It should be possible to protect RTT contents with usual means for
privacy and integrity.Ref RFC 6881 section 16. [RFC6881]
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]
The mixer performance requirements can be expressed in two
figures.
1) The number of participants who can transmit simultaneously with
the text not being delayed in the mixer more than 500
milliseconds. This requirement is depending on the application.
Five simultaneous transmitting participants is a sufficiently high
number for most situations.
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2) The switching time from when the mixer is transmitting text
from one participant and text arrives from another participant,
until the mixer sends the text from the second participant. This
time should not be more than 500 milliseconds.
4. 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. 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.
4.1.1. 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 switching time will thus be 900 milliseconds. The mixer can not
even send text from two simultaneous sources without introducing more
than 500 milliseconds delay. 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.
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4.1.2. RTP Mixer using the default method but decreased transmission
interval
This method makes use of the default RTP-mixing method briefly
described in section Section 4.1.1. 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. This
increases the switching performance to three source switches per
second. The delay of new text from a participant can be one second.
Pros:
Minor influence on standards
Cons:
Too long delay of new text from more than two simultaneous sources.
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.
4.1.3. 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.
The transmission interval should be 100 milliseconds when there is
text to transmit from more than one source.
The details for application of this method together with RFC 4103
[RFC4103] are specified in
[I-D.hellstrom-avtcore-multi-party-rtt-source]
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The identification of the sources is made through the CSRC fields and
can be made more readable through the RTCP SDES CNAME and NAME
packets as described in RTP[RFC3550].
Also information provided through the notification according to RFC
4575 [RFC4575] when the participant joined the conference provides
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.
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.
Pros:
This method has low overhead and less complexity than the methods in
sections Section 4.1.1, Section 4.1.2, Section 4.1.4 and
Section 4.1.5.
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. There can be switching between five source
per second with an introduced delay of maximum 500 ms. With just two
parties typing simultaneously, the delay will be a maximum of 100 ms.
Cons:
When more consecutive packet loss than the number of generations of
redundant data appears, it is not possible to deduct the sources of
the totally lost data.
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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4.1.4. RTP Mixer using timestamp to identify redundancy
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 merged. The recovery procedure in
the receiver will use the RTP timestamp and timestamp offsets in the
redundancy headers to evaluate if a piece of redundant data should be
recovered or not in case of packet loss.
In this method, the transmission interval is 100 milliseconds when
text from more than one source is available for transmission.
Pros:
The format of each packet is equal to what is specified in RFC 4103
[RFC4103].
The source switching performance is sufficient. 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.
Cons:
The recovery time in case of packet loss is long. With five
participants, it will be 1.5 seconds.
The recovery procedure is complex and very different from what is
described in RFC 4103 [RFC4103].
It is not sure that this change can be regarded to be an update to
RFC 4103.
4.1.5. 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 redundancy
header reflects these parts of the payload. The CSRC list contains
one CSRC member per source in the payload.
The maximum number of members in the CSRC-list is 16, 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.
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The transmission interval is set to 100 milliseconds when there is
text from more than one source available for transmission.
Pros:
The source switching performance is good. Text from five (and in
fact 16) participants can be transmitted simultaneously with 300
milliseconds interval per source.
New text from five (and in fact 16) simultaneous sources can be
transmitted within 300 milliseconds. This is good performance.
Cons:
The format of each packet is very different from what is specified in
RFC 4103 [RFC4103].
It is doubtful if this change can be seen as just an update to RFC
4103.
The recovery procedure is complex and very different from what is
described in RFC 4103 [RFC4103].
It is doubtful if this change can be regarded to be an update to RFC
4103.
4.1.6. 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
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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:
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.
Transmitted text 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 deduct 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.
The conference server need to be allowed to decrypt/encrypt the
packet payload.
4.1.7. 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
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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.
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 Appendix A for an informative example of a procedure for
presenting RTT to a conference-unaware UA.
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 and MUST
NOT be used as the main method, but only as the last resort for
backwards interoperability with multi-party unaware endpoints.
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The conference server need to be allowed to decrypt/encrypt the
packet payload.
4.2. RTP-based bridging with RTT media contents not touched 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. 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.
4.2.1. RTP Translator sending one RTT stream per participant
Within the RTP session, text from each participant is transmitted
from the RTP media translator in a separate RTP stream, thus using
the same destination address/port combination, 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 theSSRC and the RTCP
SDES CNAME and NAME packets as described in RTP[RFC3550].
Pros:
This method has moderate overhead. 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.
Cons:
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There may be RTP implementations not supporting the Translator model.
This configuration is not supported by current media declarations in
sdp. RFC 3264 [RFC3264]specifies in many places that one media
description is supposed to describe just one RTP stream.
4.2.2. 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.
4.2.3. 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
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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
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.
4.2.4. 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.
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4.3. 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.
4.3.1. 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.
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.
Cons:
With a high number of participants, the overhead of establishing the
high number of data channels required may be high.
4.3.2. 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 4.1.6.
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.
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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:
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.
5. Preferred 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 with frequent
transmission and indicating sources in the CSRC-list" in
Section 4.1.3.
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 4.2.2.
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 4.3.1.
6. Session control of 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.
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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.
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.
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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.
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.
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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.
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-mix".
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
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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-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 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.
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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 the name "rtt-mix" with 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 suppressed or found to
be possible to negotiate in another way.
An offer-answer exchange should take place and if both parties
specify "rtt-mix" capability, 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.
Can be used on specific text media.
Could possibly be used also for SDP-controlled WebRTC sessions with
multiple streams in the same data channel.
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-mix", with one or more of its possible values.
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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 cps=30;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 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:
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.
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6.6. Preferred capability declaration method.
The preferred capability declaration method is the one with a
simplified SDP attribute "a=rtt-mix" Section 6.4 because it is
straightforward and partially usable also for WebRTC.
7. Identification of the source of text
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
labeling 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.
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 what
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.
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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.
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.
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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.
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.
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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: Example of a three-party call presented in the bubble
style.
Figure 1: Three-party call with bubble style.
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.
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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 UA:s 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 UA, 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 realistic alternative is to
not allow multi-party sessions with multi-party unaware endpoints.
See Appendix A for an informative example of a procedure for
presenting RTT to a conference-unaware endpoint.
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.
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
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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. Changes
14.1. Changes from draft-hellstrom-mmusic-multi-party-rtt-02 to draft-
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]".
14.2. 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 referenes to its
spec for details.
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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.
[I-D.hellstrom-avtcore-multi-party-rtt-source]
Hellstrom, G., "Indicating source of multi-party Real-time
text", draft-hellstrom-avtcore-multi-party-rtt-source-02
(work in progress), March 2020.
[I-D.ietf-mmusic-t140-usage-data-channel]
Holmberg, C., "T.140 Real-time Text Conversation over
WebRTC Data Channels", draft-ietf-mmusic-t140-usage-data-
channel-12 (work in progress), March 2020.
[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)", draft-ietf-perc-private-media-
framework-12 (work in progress), June 2019.
[NENAi3] NENA, "NENA-STA-010.2-2016. Detailed Functional and
Interface Standards for the NENA i3 Solution", October
2016.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., 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>.
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[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>.
[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>.
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[RFC4597] Even, R. and N. Ismail, "Conferencing Scenarios",
RFC 4597, DOI 10.17487/RFC4597, August 2006,
<https://www.rfc-editor.org/info/rfc4597>.
[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>.
[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>.
[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.
[T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000),
Protocol for multimedia application text conversation",
February 2000.
[TS103479]
ETSI, "TS 103 479. Emergency communications (EMTEL); Core
elements for network independent access to emergency
services", December 2019.
[TS22173] 3GPP, "IP Multimedia Core Network Subsystem (IMS)
Multimedia Telephony Service and supplementary services;
Stage 1", 3GPP TS 22.173 17.1.0, December 2019.
[TS24147] 3GPP, "Conferencing using the IP Multimedia (IM) Core
Network (CN) subsystem; Stage 3", 3GPP TS 24.147 16.0.0,
December 2019.
Appendix A. Mixing for a multi-party unaware endpoint
This informational appendix describes media mixer procedures for a
multi-party conference server to format real-time text from a number
of participants into one single text stream to a participant with an
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endpoint that has no features for multi-party text display. The
procedures are intended for implementations using ITU-T T.140 [T.140]
for the real-time text coding and presentation.
A.1. Short description
The media mixer procedures described here are intended to make real-
time text from a number of call participants be coordinated into one
text stream to an endpoint originally intended for two-party calls.
A conference server is supposed to apply the procedures.
The procedures may also be applied on an endpoint for display of
multiple streams of real-time text in one area.
The intention is that text from each participant shall be displayed
in suitable sections so that they are easy to read, and text from one
active participant at a time is sent and displayed in real-time. The
receiving terminal is assumed to have one display area for received
text. The display is arranged by this procedure in a text chat
style, with a name label in front of each text section where switch
of source of the text has taken place.
When more than one participant transmits text at the same time, the
text from only one of them is transmitted directly to the receiving
terminals. Text from the other participants is stored in buffers in
the conference server for transmission at a later time, when a
suitable situation for switch of current transmitter can take place.
A.2. Functionality goals and drawbacks
The procedures are intended to make best efforts to present a multi-
party text conversation on an endpoint that has no awareness of
multi-party calls. There are some obvious drawbacks, and an endpoint
designed with multi-party awareness will be able to present multi-
party call contents in a more flexible way. Only two parties at a
time will be allowed to display text added in real-time, while the
other parties' produced text will need to be stored in the multi-
party server for a moment awaiting a suitable occasion to be
displayed. There are also some cases of erasure that will not be
performed on the target text but only indicated in another way. Even
with these drawbacks, the procedure provides an opportunity to
display text from more than two parties in a smooth and readable way.
This specification does not introduce any new protocol element, and
does not rely on anything else than basic two-party terminal
functionality with presentation level according to ITU-T T.140
[T.140]. It is a description of a best current practice for mixing
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and presentation of the real-time text component in multi-party calls
with terminals without multi-party awareness.
The procedures are applicable to scenarios, when the conference focus
and a User Agent have not gone through any successfully completed
negotiation about multi-party awareness for the real-time text medium
neither on the transport level, nor on the presentation level.
A.3. Definitions
Active participant: Any user sending text, or being in a pending
period.
BOM Byte-Order-Mark, the Unicode character FEFF in UCS-16.
Buffer: A buffer intended for unsent text collected per
participant.
Contributing participants: The participants selected to contribute
to the text stream sent to the recipients.
By default all participants except the recipient are contributing
participants for transmission to the recipient.
Current participant: The participant for whom text currently is
transmitted to the recipient in real time.
Current Recipients: By default all participants.
Display Counter: A counter for the number of displayable
characters in a participant's buffer or in the current entry.
Used for controlling how far erasure may be performed.
Erasure replacement: A character to be displayed when an erasure
was done, but the text to erase is not reachable on the multi-
party display. Default 'X'.
Message delimiter: Character(s) forming the end of an imagined
message. A configurable set of alternatives, consisting by
default of: Line Separator, Paragraph Separator, CR, CRLF, LF.
Pending period: A configurable time period of inactivity from a
participant, by default set to 7 seconds after each reception of
characters from that participant, evaluated as current time minus
time stamp of latest entered character.
Sentence delimiter: Characters forming end of sentence: A
configurable set of alternatives, by default consisting of: dot
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'.', question mark '?' and exclamation mark '!' followed by a
space.
Label: A readable unique name for a participant, created by the
server from a suitable source related to the participant,
surrounded by the Label delimiters. The label should have a
settable maximum length, with 12 being the default.
Label delimiters A configurable set of characters at the edges of
the Label, by default being a left bracket [ at the leading edge
and a closing bracket ] followed by a space at the trailing edge.
Line Separator Unicode UCS-16 2028. Used to request NewLine in
Real-Time Text.
Maximum waiting time: The maximum time any participant's text
shall be allowed to wait for transmission, by default set to 20
seconds.
Recipient: The terminal receiving the mixed text stream.
SGR Select Graphic Rendition, a control code to specify colours
etc.
Switch Reason: A set of reasons to switch Current Participant,
consisting of the following
-Waiting time higher for any other participant than the current
participant combined with any of the following states:
-A message delimiter was the latest transmitted item
-A sentence delimiter was the latest transmitted item
-A Pending Period has expired and still no text has been
transmitted
-The Maximum Waiting time has expired followed by a Word Delimiter
or an expired Time Extension.
Waiting time: The time the first character in queue for
transmission from a participant has been waiting in a buffer for
transmission. The granularity shall be 0.3 Seconds or finer.
Word delimiter: Character forming end of word: space
Time extension: A configurable short extension time allowed after
the Maximum waiting time during which a suitable moment for
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switching Current Participant is awaited, by default set to 7
seconds.
A.4. Presentation level procedures
The conference server applies these mixing procedures to text
transmitted to call participants who have not gone through a
completed negotiation for conference awareness in real-time text
presentation.
All the participants and the conference server use real-time text
conversation presentation coding according to ITU-T T.140 [T.140]. A
consequence is that real-time text transmissions are UTF-8 coded,
with control codes selected from ISO 6429 [ISO 6429].
The description is from the conference server point of view.
A.4.1. Structure
The real-time text mixer structure described here is supposed to be
placed in the media path so that it is implemented with one mixer per
recipient. A mixer contains buffers for temporary storage of text
intended for the recipient. Each mixer has one buffer for each
contributing participant. A set of status variables is maintained
per buffer and is used in the mixer actions. The mixer logic decides
for each moment which participant's buffer content is to be sent on
to the recipient. By default, the recipient does not contribute text
to its own mixer. Text transmitted by a participant is usually
displayed locally and it will only cause confusion if it appears also
in received text.
A.4.2. Action on reception
This description of the mixer is valid per recipient.
Text from each contributing participant is checked for a set of
characteristics on reception.
Delete BOM: BOM characters are deleted.
Insert in buffer: Resulting text is put into the contributing
participant's buffer in the receiving participant's mixer.
Maintain a display counter: For each text character that will take
a position on the receiving display, a Display Counter for each
participant is increased by one.
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There is one T.140 real-time text item that consists of two
characters, but is regarded to be a unit and therefore increase
the Display Counter with one only.That is CRLF.
Furthermore, the following control codes are regarded units that
shall not take any position on the receiving display and shall
therefore not increase the Display Counter:
0098 string 009C (SOS-ST strings)
ESC 0061 (INT)
009B Ps 006D (the SGR code, with special handling described below)
BEL (Alert in session)
See the section on control codes below for details.
Combination characters: Also note that it is possible to use
combination characters in Unicode. Such combination characters
contain more than one character part. They shall only increase
the Display Counter with one. The combination characters mainly
have components in the series 0300 - 0361 and 20D0 - 20E1.
Erasure: If the control code for erasure, BS, is received, the
following shall be done: If the Display Counter is 0, an Erasure
Replacement character, by default being "X" is inserted in the
buffer instead of the erasure, to mark that erasure was intended
in earlier transmitted entries. ( this matches traditional habits
in real-time text when participants sometimes type XXX to indicate
erasure they do not bother to make explicit). If the Display
Counter is >0, then the counter is reduced by one, and the erasure
control code BS put into the buffer.
Initial action in the session: BOM shall be sent initially to the
recipients in the beginning of the session.
Maintaining a waiting time per participant: The time that text has
been in the buffer is maintained as the waiting time for each
buffer. A granularity of 0.3 seconds is sufficient.
Storing time of reception for each character: Each character that
is stored in a buffer shall be assigned with a time stamp
indicating its time of reception. A granularity of 0.3 seconds is
sufficient. This time stamp is used for calculation of idle time
and waiting time in the evaluation of switch reasons.
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Initial assignment of the Current Participant: The first
contributing participant to send text in the session is assigned
to be the Current Participant.
Actions on assignment of a Current Participant: When a participant
becomes the Current Participant, the following initial actions
shall be performed:
1. Scanning transmissions and timers for a Switch Reason is
inactivated.
2. The Current Recipients are set so that all transmissions go to
the new set of Current Recipients (See definition).
3. A Line Separator is transmitted if the switch reason was any
other than a message delimiter.
4. The Label is transmitted
5. Any stored SGR code is transmitted
6. Scanning transmissions and timers for a Switch Reason is
activated.
7. Text in the buffer is transmitted, recalculating and setting
the waiting time for each transmitted character based on the time
of reception of next character in the buffer. If a switch occurs
during transmission from the buffer, the remaining buffer contents
is maintained and transmission can continue next time this
transmitter becomes the current participant. Any text entered
into the buffer for the current participant is after that sent to
the recipient until a Switch Reason occurs.
Actions on transmission and during the session: Transmissions are
checked for control codes to act on at transmission as described
below in the section about handling of control codes and such
actions are performed. When the scanning of transmission and
timers for a Switch Reason is active, the timers and the
transmission to the recipient is analyzed for detection if a
Switch Reason has occurred. See the definition of Switch Reasons
for details.
Actions when a Switch Reason has occurred: If a Switch Reason has
occurred, then the following actions shall be performed:
1. The Display Counter of the Current Participant is set to zero
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2. If there is an SGR code stored for the Current Participant, a
reset of SGR shall be sent by the sequence SGR 0 [009B 0000 006D].
3. A participant with the longest waiting time is assigned to be
the Current Participant, and the procedure for assignment of a
Current Participant described above is performed.
Handling of Control codes: The following control codes are
specified by ITU-T T.140. Some of them require consideration in
the conference server. Note that the codes presented here are
expressed in UCS-16, while transmission is made in UTF-8 transform
of these codes. Other sections specify procedures for handling of
specific control codes in the conference server.
BEL 0007 Bell, provides for alerting during an active session.
BS 0008 Back Space, erases the last entered character.
NEW LINE 2028 Line separator.
CR LF 000D 000A A supported, but not preferred way of requesting a
new line.
INT ESC 0061 Interrupt (used to initiate mode negotiation
procedure).
SGR 009B Ps 006D Select graphic rendition. Ps is rendition
parameters specified in ISO 6429.
SOS 0098 Start of string, used as a general protocol element
introducer, followed by a maximum 256 bytes string.
ST 009C String terminator, end of SOS string.
ESC 001B Escape - used in control strings.
Byte order mark FEFF Zero width, no break space, used for
synchronization.
Missing text mark FFFD Replacement character, marks place in
stream of possible text loss.
Code for message border, useful, but not mentioned in T.140: New
Message 2029 Paragraph separator
Handling of Graphic Rendition SGR: The following procedure shall
be followed in order to let the participants control the graphic
rendition of their entries without disturbing other participants'
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graphic rendition. The text stream sent to a recipient shall be
monitored for the SGR sequence. The latest conveyed SGR sequence
is also stored as a status variable for the recipient. If the SGR
0 code initiated from the current participant is transmitted, the
SGR storage shall be cleared.
A.5. Display examples
The following pictures are examples of the view on a participant's
display.
_________________________________________________
| 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 A1 : 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.
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_________________________________________________
| |^|
|[Alice] Hi, Alice here. | |
| | |
|[mix][Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris| |
| I thought you should be here. | |
| | |
|[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|[mix][Bob] And I on Wednesday evening. | |
| | |
|[Eve] we can have dinner and then walk | |
| | |
|[Eve] But I need to be back to | |
| the hotel by 11 because I need |-|
| |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure A2 shows a conference with a real-time multi-party text view.
Bob's text is buffered until a Current switch reason.
A.6. References for this Appendix
[T.140] ITU-T T.140 Application protocol, text conversation
(including amendment 1.)
[RFC 4103] IETF RFC 4103 RTP Payload for text conversation
[RTP] IETF RFC 3550 RTP: A Transport Protocol for Real-Time
Applications.
[RFC 4579] IETF RFC 4579 SIP Call Control : Conferencing for user
agents.
[ISO 6429] ISO 6429 Control functions for coded character sets.
[UTF-8] IETF RFC 3629 UTF-8, a transformation format of ISO 10646
[Unicode] The Unicode Consortium, "The Unicode Standard ; Version
4.0.
[ISO 10646-1] ISO 10646 Universal multiple-octet coded character
set (UCS)
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[UCS-16] See ISO 10646-1
A.7. Acknowledgement for the appendix
This appendix was developed with funding in part from the National
Institute on Disability and Rehabilitation Research, U.S. Department
of Education, RERC on Telecommunications Access, grant # H133E090001.
However, the contents do not necessarily represent the policy of the
Department of Education, and you should not assume endorsement by the
Federal Government.
Author's Address
Gunnar Hellstrom
Omnitor
Esplanaden 30
Vendelso SE-136 70
SE
Phone: +46 708 204 288
Email: gunnar.hellstrom@omnitor.se
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