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draft-hellstrom-multi-party-rtt.xml
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draft-hellstrom-multi-party-rtt.xml
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<?xml version="1.0" encoding="US-ASCII"?>
<!-- This template is for creating an Internet Draft using xml2rfc,
which is available here: http://xml.resource.org. -->
<!DOCTYPE rfc SYSTEM "rfc2629.dtd" [
<!-- One method to get references from the online citation libraries.
There has to be one entity for each item to be referenced.
An alternate method (rfc include) is described in the references. -->
<!ENTITY RFC2119 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml">
<!ENTITY RFC3261 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.3261.xml">
<!ENTITY RFC3550 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml">
<!ENTITY RFC4103 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.4103.xml">
<!ENTITY RFC4353 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.4353.xml">
<!ENTITY RFC4575 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.4575.xml">
<!ENTITY RFC4579 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.4579.xml">
<!ENTITY RFC4597 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.4597.xml">
<!ENTITY RFC7667 SYSTEM "http://xml.resource.org/public/rfc/bibxml/reference.RFC.7667.xml">
]>
<?xml-stylesheet type='text/xsl' href='rfc2629.xslt' ?>
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please see http://xml.resource.org/authoring/README.html. -->
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(Here they are set differently than their defaults in xml2rfc v1.32) -->
<?rfc strict="yes" ?>
<!-- give errors regarding ID-nits and DTD validation -->
<!-- control the table of contents (ToC) -->
<?rfc toc="yes"?>
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<?rfc tocdepth="4"?>
<!-- the number of levels of subsections in ToC. default: 3 -->
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<?rfc symrefs="yes"?>
<!-- use symbolic references tags, i.e, [RFC2119] instead of [1] -->
<?rfc sortrefs="yes" ?>
<!-- sort the reference entries alphabetically -->
<!-- control vertical white space
(using these PIs as follows is recommended by the RFC Editor) -->
<?rfc compact="yes" ?>
<!-- do not start each main section on a new page -->
<?rfc subcompact="no" ?>
<!-- keep one blank line between list items -->
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<rfc category="bcp" docName="draft-hellstrom-multi-party-rtt-00"
ipr="trust200902">
<!-- category values: std, bcp, info, exp, and historic
ipr values: trust200902, noModificationTrust200902, noDerivativesTrust200902,
or pre5378Trust200902
you can add the attributes updates="NNNN" and obsoletes="NNNN"
they will automatically be output with "(if approved)" -->
<!-- ***** FRONT MATTER ***** -->
<front>
<!-- The abbreviated title is used in the page header - it is only necessary if the
full title is longer than 39 characters -->
<title abbrev="Real-time text multi-party handling">Real-time text media handling in
multi-party conferences</title>
<!-- add 'role="editor"' below for the editors if appropriate -->
<!-- Another author who claims to be an editor -->
<author fullname="Gunnar Hellstrom" initials="G." surname="Hellstrom">
<organization>Omnitor</organization>
<address>
<postal>
<street>Esplanaden 30</street>
<!-- Reorder these if your country does things differently -->
<city>Vendelso</city>
<code>SE-136 70</code>
<country>SE</country>
</postal>
<phone>+46 708 204 288</phone>
<email>[email protected]</email>
<uri>www.omnitor.se</uri>
<!-- uri and facsimile elements may also be added -->
</address>
</author>
<date month="October" year="2019" />
<!-- If the month and year are both specified and are the current ones, xml2rfc will fill
in the current day for you. If only the current year is specified, xml2rfc will fill
in the current day and month for you. If the year is not the current one, it is
necessary to specify at least a month (xml2rfc assumes day="1" if not specified for the
purpose of calculating the expiry date). With drafts it is normally sufficient to
specify just the year. -->
<!-- Meta-data Declarations -->
<area>General</area>
<workgroup>Internet Engineering Task Force</workgroup>
<!-- WG name at the upperleft corner of the doc,
IETF is fine for individual submissions.
If this element is not present, the default is "Network Working Group",
which is used by the RFC Editor as a nod to the history of the IETF. -->
<keyword>Internet-Draft</keyword>
<!-- Keywords will be incorporated into HTML output
files in a meta tag but they have no effect on text or nroff
output. If you submit your draft to the RFC Editor, the
keywords will be used for the search engine. -->
<abstract>
<t>This memo specifies methods for Real-Time Text (RTT) media handling in multi-party
calls. The main solution is to carry Real-Time text by the RTP protocol
in a time-sampled mode according to RFC 4103. The main solution for centralized
multi-party handling of real-time text is achieved through a media
control unit coordinating multiple RTP text streams into one RTP session.</t>
<t>
Identification for the streams are provided through the RTCP messages.
This mechanism enables the receiving application to present the received
real-time text medium in different ways according to user preferences.
Some presentation related features are also described explaining
suitable variations of transmission and presentation of text.
</t>
<t>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 one is provided. Two 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.</t>
<t>Brief information is also provided for multi-party RTT in the WebRTC environment.</t>
<t>EDITOR NOTE: A number of alternatives are specified for discussion. A
decision is needed which alternatives are preferred and then how
the preferred alternatives shall be emphasized.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>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.</t>
<t>Media from a number of multimedia session participants can be combined
in a multi-party session. This memo specifies how the real-time text streams
are handled in multi-party sessions.</t>
<t>The description is mainly focused on the transport level, but also
describes a few session and presentation level aspects.</t>
<t>Transport of real-time text is specified in <xref
target="RFC4103">RFC 4103</xref> RTP Payload for text conversation. It
makes use of <xref target="RFC3550">RFC 3550</xref> Real Time Protocol,
for transport. Robustness against network transmission problems is normally
achieved through redundancy transmission based on the principle from RFC 2198,
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
<xref target="RFC3261">RFC 3261</xref> environment.</t>
<t>A very brief overview of functions for real-time
text handling in multi-party sessions is described in <xref
target="RFC4597">RFC 4597</xref> Conferencing Scenarios, sections 4.8 and 4.10. This
specification builds on that description and indicates which
protocol mechanisms should be used to implement multi-party handling of
real-time text.</t>
<t>EDITOR NOTE: A number of alternatives are specified for discussion. A
decision is needed which alternatives are preferred and then how
the preferred alternatives shall be emphasized.</t>
<section title="Requirements Language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
</section>
<section title="Centralized conference model">
<t>In the centralized conference model for SIP, introduced in <xref
target="RFC4353">RFC 4353</xref> 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 medium.</t>
<t>The central function is called the Focus UA and may be co-located in
an advanced terminal including multi-party control functions, or it may
be located in a separate location. 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.</t>
<t>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.</t>
<t>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. <xref
target="RFC7667">RFC 7667</xref> 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.</t>
</section>
<section title="Requirements on multi-party RTT">
<t>The following requirements are placed on multi-party RTT:</t>
<t>
<list style="empty">
<t>The solution shall be applicable to IMS (3GPP TS 22.173), SIP based
VoIP and Next Generation Emergency Services (NENA i3, EENA NG LTD, RFC 6443). </t>
<t>The transmission interval for text must not be longer than 500 milliseconds when
there is anything available to send. Ref ITU-T T.140.</t>
<t>If text loss is detected or suspected, a missing text marker shall be inserted in
the text stream where the loss is detected or suspected. Ref ITU-T T.140 Amendment 1.
ETSI EN 301 549</t>
<t>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 </t>
<t>Bridges must be multimedia capable (voice, video, text). Ref NENA i3 STA-010.2. </t>
<t>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. </t>
<t>It should be possible to protext RTT contents with usual means for privacy and integrity.Ref RFC 6881 section 16</t>
<t>Conferencing procedures are documented in RFC 4579. Ref NENA i3 STA-010.2. </t>
<t>Conferencing applies to any kind of media stream by which users may want to communicate...
Ref 3GPP TS 24.147</t>
<t>The framework for SIP conferences is specified in RFC 4353. Ref 3GPP TS 24.147</t>
</list>
</t>
</section>
<section title="Coordination of text RTP streams">
<t>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.</t>
<section title="RTP Translator sending one RTT stream per participant,">
<t>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 <xref target="RFC3550">RFC
3550</xref> about the Translator function. The sources of the text
in each RTP packet are identified by the SSRC parameters in the RTP packets, containing the
SSRC of the initial sources of text. This method is described in RFC 7667,
section 3.5.1 Relay-transport translator or 3.5.2 Media translator.</t>
<t>The identification of the source is made through the RTCP SDES
CNAME and NAME packets as described in RTP<xref
target="RFC3550"></xref>.</t>
<t>Pros:</t>
<t> 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 stream. (normally primary and
two redundant levels. </t>
<t>More loss than what can be recovered, can be detected and the marker for text
loss can be inserted in the correct stream.</t>
<t>It may be possible in some scenarios to keep the text encrypted through the Translator.</t>
<t>Cons:</t>
<t>There may be RTP implementations not supporting the Translator model. </t>
<t>It is even most likely that this configuration is not supported by current media declarations in sdp.
RFC 3264 specifies in many places that one media description is supposed to describe just one RTP stream. </t>
</section>
<section title="RTP Mixer indicating participants in CSRC">
<t>
An RTP media mixer combines text from all participants except from the receiving endpoint 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 <xref target="RFC3550">RFC
3550</xref> about the Mixer function. The sources of the text
in each RTP packet are identified by the CSRC parameters in the RTP packets, containing the
SSRC of the initial sources of text. The order of the CSRC parameters are the
same as the order of the redundant and primary data fields in the packet. If all redundancy
blocks in a packet are from the same source,
then it is allowed to use only one CSRC in the RTP packet. This method is described in RFC 7667,
section 3.6.3 Media switching mixer.</t>
<t>The identification of the source is made through the RTCP SDES
CNAME and NAME packets as described in RTP<xref target="RFC3550"></xref>.</t>
<t>It is likely that the conference server need to have authority to decrypt the payload in the 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. Further study is needed</t>
<t>Pros:</t>
<t> This method has moderate overhead. </t>
<t>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 stream. (normally primary and
two redundant levels.</t>
<t>This method can be implemented with most RTP implementations.</t>
<t></t>
<t>Cons:</t>
<t> When 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.
Therefore it is not possible to know in which stream to insert the missing text marker.
It MAY be acceptable to either indicate a general loss indication, or insert a loss marker in all streams.
Calculations of most likely source can however be made from received RTP and RTCP
contents so that the loss marker can be inserted in the most likely struck stream.</t>
<t>The conference server need to be allowed to decrypt/encrypt the packet payload.</t>
</section>
<section title="RTP Mixer indicating participants by a control code in the stream">
<t>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 <xref target="RFC3550">RFC
3550</xref> 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.</t>
<t>The receiver can use the string for presenting the source of text. This method is
on the RTP level described in RFC 7667, section 3.6.2 Media mixing mixer.</t>
<t>The inline coding of the source of text is applied in the data stream itself, and an
RTP mixer function normally without CSRC identification used for
coordinating the sources of text into one RTP stream.</t>
<t>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 <xref target="T.140">ITU-T T.140</xref>.
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.</t>
<t> A receiving UA SHOULD separate text from the different sources and
identify and display them accordingly.</t>
<t>The conference server need to be allowed to decrypt/encrypt the packet payload in order to check the source and repack the text.</t>
<t>Pros: </t>
<t>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 stream. (normally primary and
two redundant levels.</t>
<t>This method can be implemented with most RTP implementations.</t>
<t>Transmitted text can also be used with other transports than RTP</t>
<t>Cons: </t>
<t>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.
Therefore it is not possible to know in which stream to insert the missing text marker.
Calculations of most likely source can however be made from recent history, so that it is quite likely that the marker is inserted in the correct stream. Such loss should however be rare, and a general warning that there might have been text loss in the session might be acceptable.</t>
<t>The mixer needs to be able to generate suitable and unique source identifications
which are suitable as labels for the sources. </t>
<t>Requires an extension on the ITU-T T.140 standard, best made by the ITU. </t>
<t>The conference server need to be allowed to decrypt/encrypt the packet payload.</t>
<t>The conference server need to be allowed to decrypt/encrypt the packet payload.</t>
</section>
<section title="Mesh of RTP endpoints">
<t>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. </t>
<t>This method is described in RFC 7667, section 3.4 Point to multi-point using mesh.</t>
<t>Pros: </t>
<t>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 stream. (normally primary and two redundant levels.</t>
<t>This method can be implemented with most RTP implementations.</t>
<t>Transmitted text can also be used with other transports than RTP</t>
<t>Cons: </t>
<t>This model is not described in IMS, NENA and EENA specifications, and does therefore not meet the requirements.</t>
</section>
<section title="Multiple RTP sessions, one for each participant">
<t>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. </t>
<t>This method is out of scope for further discussion here, because the foreseen applications use centralized model conferencing.</t>
<t>Pros: </t>
<t>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 stream. (normally primary and two redundant levels.</t>
<t>Complete loss of text can be indicated in the received stream.</t>
<t>This method can be implemented with most RTP implementations.</t>
<t>End-to-end encryption is achievable.</t>
<t>Cons:</t>
<t> This method is not described in IMS, NENA and EENA specifications and does therefore not meet the requirements.</t>
<t>A lot of network resources are spent on setting up separate sessions for each participant.</t>
</section>
<section title="Mixing for conference-unaware user agents">
<t>Multi-party real-time text contents can be transmitted to
conference-unaware user agents if source labeling 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 trhe 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.</t>
<t>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 3.6.2 Media mixing mixer.</t>
<t> See Appendix A for an informative example of a procedure for presenting RTT to a conference-unaware UA.</t>
<t>Pros:</t>
<t> Can be transmitted to conference-unaware endpoints.</t>
<t>Can be used with other transports than RTP</t>
<t>Cons:</t>
<t> Does not allow full real-time presentation of more than one source at a time. Text from other sources will
be delayed, even if automatic detection of suitable moments for switching source for presentation is made by the mixer.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>Because of all these cons, this method MUST NOT be used as the main method, but only as the last resort
for backwards interoperability with conference-unaware endpoints. </t>
<t>The conference server need to be allowed to decrypt/encrypt the packet payload.</t>
</section>
</section>
<section title="RTT bridging in WebRTC">
<t>Within WebRTC, real-time text is specified to be carried in WebRTC data
channels as specified in draft-ietf-mmusic-t140-usage-data-channel. A few ways to handle multi-party
RTT are mentioned briefly. They are explained and further detailed below.</t>
<section title="RTT bridging in WebRTC with one data channel per source">
<t>
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.
</t>
<t>The stream-id forms a unique stream identification.</t>
<t>The identification of the source is made through the Label property of the channel,
and session information belonging to the source. The UA can compose a readable label for the presentation from this information.</t>
<t>Pros: </t>
<t> This is a straightforward solution. </t>
<t>Cons: </t>
<t>With a high number of participants, the overhead of establishing the high number of data channels required may be high.</t>
</section>
<section title="RTT bridging in WebRTC with one common data channel">
<t>
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.</t>
<t>This method is described in a corresponding section for RTP transmission above. </t>
<t>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 <xref target="T.140">ITU-T T.140</xref>).</t>
<t>The UA 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.</t>
<t>Pros:</t>
<t> This solution has relatively low overhead on session and network level </t>
<t>Cons:</t>
<t> This solution has higher overhead on the media contents level than the WebRTC solution above.</t>
<t>Standardisation of the new control code "c" in ITU-T T.140 is required.</t>
<t>The conference server need to be allowed to decrypt/encrypt the data channel contents.</t>
</section>
</section>
<section title="Preferred multi-party RTT transport method">
<t>EDITOR NOTE: The recommendations here need to be validated, and the proposed further studies performed.</t>
<t>For RTP transport of RTT, two methods for multi-party mixing and transport for conference-aware parties stand out as fulfilling the goals best: "RTP Mixer indicating participants in CSRC" and "RTP Mixer indicating participants by a control code in the stream". The CSRC based method has a slightly better opportunity to use a robust and well defined procedure in the server. The inline stream based method has the slightly better opportunity for ease of interworking with other environments for RTT where the in-line identification also could be used. The robustness is important, and therefore the method to implement for multi-party RTT with conference-aware parties when no other method is explicitly agreed between implementing parties for SIP with RTP is "RTP Mixer indicating participants in CSRC".</t>
<t>Further studies should be made to find out if assessment of the source for lost text can be better done, and if operation without letting the conference server decrypt data can be specified.</t>
<t></t>
<t>For WebRTC, one method is to prefer because it seems to allow the conference server to act on the text without decrypting it. So, for WebRTC, the method to implement for multi-party RTT with conference-aware parties when no other method is explicitly agreed between
implementing parties is: "RTT bridging in the WebRTC environment with one data channel per source".</t>
<t>Further studies are needed to check the encryption statement above.</t>
<t>As a last resort, when the UA is not conference-aware, the method for mixing for conference-unaware user agents may be used for both RTP and WebRTC data channel solutions considering that this method provides a reduced impression of the real time characteristics and may delay presentation of text.</t>
</section>
<section title="Session control of multi-party RTT sessions">
<t>General session control aspects for multi-party sessions are
described in <xref target="RFC4575">RFC 4575</xref> A Session
Initiation Protocol (SIP) Event Package for Conference State, and
<xref target="RFC4579">RFC 4579</xref> Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents. The nomenclature of
these specifications are used here.</t>
<t>The procedures for a conference-aware model for RTT-transmission shall only be applied if a
capability exchange for conference-aware real-time text transmission has
been completed and a supported method for multi-party real-time text transmission can be identified.</t>
<t>A method for detection of conference-awareness for centralized SIP conferencing in general is
specified in <xref target="RFC4579">RFC 4579</xref>. The focus sends the "isfocus" feature tag in a SIP Contact header. This causes
the conference-aware UA to subscribe to conference notifications from the focus. The focus then sends
notifications to the UA 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. The mechanism is described in detail in <xref target="RFC4575">RFC 4575</xref>. </t>
<t>Before a conference media server starts sending multi-party RTT to a UA, 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 according to this specification.</t>
<section title="Implicit RTT multi-party capability indication">
<t>
Capability for RTT multi-party handling can be decided to be implicitly indicated by session control items. </t>
<t>The focus may implicitly indicate muti-party RTT capability by including the media child with value "text"
in the RFC 4575 conference-info provided in conference notifications.</t>
<t>A UA 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. </t>
<t>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.</t>
<t>The implicit RTT capability indication means for the UA that it can handle multi-party RTT according to
the preferred method indicated in the RTT multi-party methods section above.</t>
<t>If the focus detects that a UA implicitly declared RTT multi-party capability, it SHALL provide RTT
according to the preferred method.</t>
<t>If the focus detects that the UA does not indicate any RTT multi-party capability, then it shall provide
RTT multi-party text in the way specified for conference-unaware UA above.</t>
<t>If the UA 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.</t>
<t>Pros: </t>
<t>Acceptance of implicit multi-party capability implies that no standardisation of explicit RTT
multi-party capability exchange is required.</t>
<t>Cons: </t>
<t>There may be a desire to indicate conference-awareness in general, but not for RTT.
Then the method called "Mixing for conference-unaware user agents" should be used as a lower functionality fallback.
There is no way to provide that indication by the UA according to the specification of the implicit method above.
The solution must be that no conference awareness is indicated by the UA when it has no RTT multi-party capability.</t>
<t>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. </t>
</section>
<section title="RTT multi-party capability declared by SIP media-tags">
<t>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 and RFC 3841.
Capability for the RTT Multi-party capability is then indicated by the media feature tag "rtt-mixer", with one
or more of its possible values in a comma-separated list.</t>
<t>The possible values in the list are:</t>
<t>
<list style="empty">
<t>rtp-translator</t>
<t>rtp-mixer</t>
<t>t140-mixer</t>
<t>rtp-mesh</t>
<t>multi-session</t>
</list></t>
<t>rtp-translator indicates capability for using the RTP-translator
based coordination of multi-party text.</t>
<t>rtp-mixer indicates capability for using the RTP-mixer based
presentation of multi-party text.</t>
<t> t140-mixer indicates capability for
using the T.140 control code source indicators in a mixer. </t>
<t>text-mixer
indicates capability for using the fallback method with text
formatting for conference-unaware endpoints.</t>
<t>rtp-mesh indicates capability for using the mesh
based transmission of multi-party text.
</t>
<t>multi-session indicates capability for using separate point-to-point
RTP sessions between all participants.</t>
<t></t>
<t>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.</t>
<t>When no common method is declared, then only the fallback method can be used.</t>
<t>If more than one text media line is included in SDP, all must be capable of using the declared RTT multi-party method.</t>
<t>Pros:</t>
<t> Provides a clear decision method.</t>
<t>Can be extended with new mixing methods.</t>
<t>Can guide call routing to a suitable capable focus.</t>
<t>Cons:</t>
<t> Requires standardization and IANA registration.</t>
<t>Cannot be used in the WebRTC environment.</t>
</section>
<section title="SDP media attribute for RTT multi-party capability indication">
<t>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-mixer", with one
or more of its possible values in a comma-separated list.</t>
<t>The possible values in the list are:</t>
<t>
<list style="empty">
<t>rtp-translator</t>
<t>rtp-mixer</t>
<t>t140-mixer</t>
<t>rtp-mesh</t>
<t>multi-session</t>
</list></t>
<t>rtp-translator indicates capability for using the RTP-translator
based coordination of multi-party text.</t>
<t>rtp-mixer indicates capability for using the RTP-mixer based
presentation of multi-party text.</t>
<t> t140-mixer indicates capability for
using the T.140 control code source indicators in a mixer. </t>
<t>text-mixer
indicates capability for using the fallback method with text
formatting for conference-unaware endpoints.</t>
<t>rtp-mesh indicates capability for using the mesh
based transmission of multi-party text.
</t>
<t>multi-session indicates capability for using separate point-to-point
RTP sessions between all participants.</t>
<t></t>
<t>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.</t>
<t>When no common method is declared, then only the fallback method can be used.</t>
<t>Pros:</t>
<t> Provides a clear decision method.</t>
<t>Can be extended with new mixing methods.</t>
<t>Can be used on specific text media.</t>
<t>Can be used also for SDP-controlled WebRTC sessions with multiple streams in the same data channel.</t>
<t>Cons:</t>
<t> Requires standardization and IANA registration.</t>
<t>Is not well defined for multi-party methods involving more than one media section for text.</t>
<t>Cannot guide SIP routing.</t>
</section>
<section title="Preferred capability declaration method.">
<t>The preferred capability declaration method is the one with SDP attributes because it is partially usable also for WebRTC.
</t>
</section>
</section>
<section title="Identification of the source of text">
<t>EDITOR NOTE: The text in the following sections need to be adapted after recommendations for the main methods for coordination of RTT has been selected. Details should be provided mainly for the recommended method.</t>
<t>As soon as a new member is added to the RTP session, its
characteristics shall be transmitted in RTCP SDES CNAME and NAME reports
according to section 6.5 in RFC 3550. The information about the participant
MUST also be included in the conference data including the text media member
in a notification according to RFC 4575.</t>
<t>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.</t>
<t>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.</t>
</section>
<section title="Presentation of multi-party text">
<t>All session participants 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.</t>
<section title="Associating identities with text streams">
<t>A source identity SHOULD be composed from available information
sources and displayed together with the text as indicated in ITU-T
T.140 Appendix<xref target="T.140"> </xref>.</t>
<t>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.</t>
<t>Applications may abbreviate the presented source identity to a
suitable form for the available display.</t>
</section>
<section title="Presentation details for multi-party aware UAs.">
<t>TBD</t>
</section>
<section title="Presentation details for multi-party unaware UAs.">
<t>TBD</t>
<t> See Appendix A for an informative example of a procedure for presenting RTT to a conference-unaware UA.</t>
</section>
</section>
<section title="Transmission of text from each user">
<t>UAs participating in sessions with real-time text, SHOULD send SDES
packets in RTCP giving values to appropriate identification fields.</t>
<t>The CNAME field SHALL be included in SDES packets.</t>
<t>The NAME field should be given a value that is suitable as an
identifier of text from the user of the UA.</t>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>EDITOR NOTE: TBD after decision of proposed preferences in the draft.</t>
<t>This document Introduces the TBD /SIP media tag/SDP media level attribute/ rtt-mixer, with a
comma-separated parameter list containing the following possible
values:</t>
<t>
<list style="empty">
<t>rtp-translator</t>
<t>rtp-mixer</t>
<t>t140-mixer</t>
<t>rtp-mesh</t>
<t>multi-session</t>
</list>
</t>
<t>rtp-translator indicates capability for using the RTP-translator
based coordination of multi-party text.</t>
<t>rtp-mixer indicates capability for using the RTP-mixer based
presentation of multi-party text.</t>
<t> t140-mixer indicates capability for
using the T.140 control code source indicators in a mixer. </t>
<t>text-mixer
indicates capability for using the fallback method with text
formatting for conference-unaware endpoints.</t>
<t>rtp-mesh indicates capability for using the mesh
based transmission of multi-party text.
</t>
<t>multi-session indicates capability for using separate point-to-point
RTP sessions between all participants.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>The security considerations valid for RFC 4103 and RFC 3550 are valid
also for the multi-party sessions with text.</t>
</section>
<section title="Congestion considerations">
<t>The congestion considerations described in RFC 4103 are valid also
for multi-party use of the real-time text RTP transport. A risk for
congestion may appear if a number of conference participants are active
transmitting text simultaneously, because this multi-party transmission
method does not allow multiple sources of text to contribute to the same
packet.</t>
<t>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.</t>
</section>
<section title="Acknowledgements">
<t>Arnoud van Wijk for contributions to an earlier, expired draft of this memo.</t>
</section>
</middle>
<!-- *****BACK MATTER ***** -->
<back>
<!-- References split into informative and normative -->
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2. simply use a PI "less than character"?rfc include="reference.RFC.2119.xml"?> here
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<references title="Normative References">
<!--?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml"?-->
<!-- <?rfc include="http://xml.resource.org/public/rfc/bibxml3/reference.I-D.hellstrom-textpreview.xml"?> -->
&RFC2119;
&RFC3261;
&RFC3550;
&RFC4103;
&RFC4575;
&RFC4579;
<reference anchor="T.140" target="http://www.itu.int/rec/T-REC-T.140/en">
<front>
<title>Protocol for multimedia application text conversation</title>
<author surname="ITU-T">
<organization></organization>
</author>
<date year="1998" />
</front>
</reference>
</references>
<references title="Informative References">
&RFC4353;
&RFC4597;
&RFC7667;
</references>
<section anchor="app-unaware" title="Mixing for a conference-unaware UA">
<t>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 a terminal 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.</t>
<section title="Short description">
<t>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 a terminal originally intended for two-party calls.
A conference server is supposed to apply the procedures.</t>
<t>The procedures may also be applied on a terminal for display of multiple streams of real-time text in one area.</t>
<t>The intention is that text from each participant shall be displayed in suitable sections so that it is 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.</t>
<t>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.</t>
</section>
<section title="Functionality goals and drawbacks">
<t>The procedures are intended to make best efforts to present a multi-party text conversation on a terminal that has no awareness of multi-party calls. There are some obvious drawbacks, and a terminal 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 added text 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.</t>
<t>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 and presentation of the real-time text component in multi-party calls with terminals without multi-party awareness.</t>
<t>The procedures are applicable to scenarios, when the conference focus and a User Agent have not gone through any successfully completed negotiation about conference awareness for the real-time text medium neither on the transport level, nor on the presentation level.</t>
</section>
<section title="Definitions">
<t>
<list style="empty">
<t>Active participant: Any user sending text, or being in a pending period.</t>
<t>BOM Byte-Order-Mark, the Unicode character FEFF in UCS-16.</t>
<t>Buffer: A buffer intended for unsent text collected per participant.</t>
<t>Contributing participants: The participants selected to contribute to the text stream sent to the recipients.</t>
<t>By default all participants except the recipient are contributing participants for transmission to the recipient.</t>
<t>Current participant: The participant for whom text currently is transmitted to the recipient in real time.</t>
<t>Current Recipients: By default all participants.</t>
<t>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.</t>
<t>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?.</t>
<t>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.</t>
<t>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.</t>
<t>Sentence delimiter: Characters forming end of sentence: A configurable set of alternatives, by default consisting of: dot ?.?, question mark ??? and exclamation mark ?!? followed by a space.</t>
<t>Label: A readable unique name for a participant, created by the server from a suitable source related to the participant, e.g. part of the SIP Display name, surrounded by the Label delimiters. The label should have a settable maximum length, with 12 being the default.</t>
<t>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.</t>
<t>Line Separator Unicode UCS-16 2028. Used to request NewLine in Real-Time Text.</t>
<t>Maximum waiting time: The maximum time any participant?s text shall be allowed to wait for transmission, by default set to 20 seconds.</t>
<t>Recipient: The terminal receiving the mixed text stream.</t>
<t>SGR Select Graphic Rendition, a control code to specify colours etc.</t>
<t>Switch Reason: A set of reasons to switch Current Participant, consisting of the following</t>
<t>-Waiting time higher for any other participant than the current participant combined with any of the following states:</t>
<t>-A message delimiter was the latest transmitted item</t>
<t>-A sentence delimiter was the latest transmitted item</t>
<t>-A Pending Period has expired and still no text has been transmitted</t>
<t>-The Maximum Waiting time has expired followed by a Word Delimiter or an expired Time Extension.</t>
<t>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.</t>
<t>Word delimiter: Character forming end of word: space </t>
<t>Time extension: A configurable short extension time allowed after the Maximum waiting time during which a suitable moment for switching Current Participant is awaited, by default set to 7 seconds. </t>
</list>
</t>
</section>
<section title="Presentation level procedures">
<t>The conference server applies these mixing procedures to text transmitted to all call participants who have not gone through a completed negotiation for conference awareness in real-time text presentation.</t>
<t>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].</t>
<t>The description is from the conference server point of view.</t>
<section title="Structure">
<t>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 will only cause confusion if it appears also in received text.</t>
<t>If there is a reason, own text can be configured to be transmitted also to the participants. That can enable a simplification of the mixer design to have only one common set of buffers instead of a set per recipient. That simplification will however hamper the flow of the conversation severely and is therefore NOT RECOMMENDED.</t>
</section>
<section title="Action on reception">
<t>
This description of the mixer is valid per recipient.</t>
<t>Text from each contributing participant is checked for a set of characteristics on reception.</t>
<t>
<list style="empty">
<t>Delete BOM: BOM characters are deleted.</t>
<t>Insert in buffer: Resulting text is put into the contributing participant?s buffer in the receiving participant?s mixer.</t>
<t>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.</t>
<t>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.</t>
<t>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:</t>
<t>0098 string 009C (SOS-ST strings)</t>
<t>ESC 0061 (INT)</t>
<t>009B Ps 006D (the SGR code, with special handling described below)</t>
<t>BEL (Alert in session)</t>
<t>See the section on control codes below for details.</t>
<t>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.</t>
<t>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.</t>
<t>Initial action in the session: BOM shall be sent initially to the recipients in the beginning of the session.</t>
<t>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.</t>
<t>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.</t>
<t>Initial assignment of the Current Participant: The first contributing participant to send text in the session is assigned to be the Current Participant.</t>
<t>Actions on assignment of a Current Participant: When a participant becomes the Current Participant, the following initial actions shall be performed:</t>
<t>1. Scanning transmissions and timers for a Switch Reason is inactivated.</t>
<t>2. The Current Recipients are set so that all transmissions go to the new set of Current Recipients (See definition).</t>
<t>3. A Line Separator is transmitted if the switch reason was any other than a message delimiter.</t>
<t>4. The Label is transmitted</t>
<t>5. Any stored SGR code is transmitted</t>
<t>6. Scanning transmissions and timers for a Switch Reason is activated.</t>
<t>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.</t>
<t>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.</t>
<t>Actions when a Switch Reason has occurred: If a Switch Reason has occurred, then the following actions shall be performed:</t>
<t>1. The Display Counter of the Current Participant is set to zero</t>
<t>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].</t>
<t>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.</t>
<t>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.</t>
<t>BEL 0007 Bell, provides for alerting during an active session.</t>
<t>BS 0008 Back Space, erases the last entered character.</t>
<t>NEW LINE 2028 Line separator.</t>
<t>CR LF 000D 000A A supported, but not preferred way of requesting a new line.</t>
<t>INT ESC 0061 Interrupt (used to initiate mode negotiation procedure).</t>
<t>SGR 009B Ps 006D Select graphic rendition. Ps is rendition parameters specified in ISO 6429.</t>
<t>SOS 0098 Start of string, used as a general protocol element introducer,
followed by a maximum 256 bytes string.</t>
<t>ST 009C String terminator, end of SOS string.</t>
<t>ESC 001B Escape - used in control strings.</t>
<t>Byte order mark FEFF Zero width, no break space, used for synchronization.</t>
<t>Missing text mark FFFD Replacement character, marks place in stream of possible text loss.</t>
<t>Code for message border, useful, but not mentioned in T.140: New Message 2029 Paragraph separator</t>
<t>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? 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.</t>
</list>
</t>
</section>
</section>
<section title="Display examples">
<t>The following pictures are examples of the view on a participant's display.</t>
<figure>
<preamble></preamble>
<artwork><![CDATA[
_________________________________________________
| 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 |
|________________________|_________________________|
]]></artwork>
<postamble>Figure 2 : 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.</postamble>
</figure>
<figure>
<artwork><![CDATA[
_________________________________________________
| |^|
|[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.| |