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Keywords: conference, bridge, SIP

Internet Engineering Task Force (IETF) G. Hellström
Request for Comments: 9071 GHAccess
Updates: 4103 July 2021
Category: Standards Track
ISSN: 2070-1721
RTP-Mixer Formatting of Multiparty Real-Time Text
Abstract
This document provides enhancements of real-time text (as specified
in RFC 4103) suitable for mixing in a centralized conference model,
enabling source identification and rapidly interleaved transmission
of text from different sources. The intended use is for real-time
text mixers and participant endpoints capable of providing an
efficient presentation or other treatment of a multiparty real-time
text session. The specified mechanism builds on the standard use of
the Contributing Source (CSRC) list in the Real-time Transport
Protocol (RTP) packet for source identification. The method makes
use of the same "text/t140" and "text/red" formats as for two-party
sessions.
Solutions using multiple RTP streams in the same RTP session are
briefly mentioned, as they could have some benefits over the RTP-
mixer model. The RTP-mixer model was selected to be used for the
fully specified solution in this document because it can be applied
to a wide range of existing RTP implementations.
A capability exchange is specified so that it can be verified that a
mixer and a participant can handle the multiparty-coded real-time
text stream using the RTP-mixer method. The capability is indicated
by the use of a Session Description Protocol (SDP) (RFC 8866) media
attribute, "rtt-mixer".
This document updates RFC 4103 ("RTP Payload for Text Conversation").
A specification for how a mixer can format text for the case when the
endpoint is not multiparty aware is also provided.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc9071.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
1.1. Terminology
1.2. Main Method, Fallback Method, and Considered Alternatives
1.3. Intended Application
2. Overview of the Two Specified Solutions and Selection of Method
2.1. The RTP-Mixer-Based Solution for Multiparty-Aware Endpoints
2.2. Mixing for Multiparty-Unaware Endpoints
2.3. Offer/Answer Considerations
2.4. Actions Depending on Capability Negotiation Result
3. Details for the RTP-Mixer-Based Mixing Method for
Multiparty-Aware Endpoints
3.1. Use of Fields in the RTP Packets
3.2. Initial Transmission of a BOM Character
3.3. Keep-Alive
3.4. Transmission Interval
3.5. Only One Source per Packet
3.6. Do Not Send Received Text to the Originating Source
3.7. Clean Incoming Text
3.8. Principles of Redundant Transmission
3.9. Text Placement in Packets
3.10. Empty T140blocks
3.11. Creation of the Redundancy
3.12. Timer Offset Fields
3.13. Other RTP Header Fields
3.14. Pause in Transmission
3.15. RTCP Considerations
3.16. Reception of Multiparty Contents
3.17. Performance Considerations
3.18. Security for Session Control and Media
3.19. SDP Offer/Answer Examples
3.20. Packet Sequence Example from Interleaved Transmission
3.21. Maximum Character Rate "cps" Setting
4. Presentation-Level Considerations
4.1. Presentation by Multiparty-Aware Endpoints
4.2. Multiparty Mixing for Multiparty-Unaware Endpoints
5. Relationship to Conference Control
5.1. Use with SIP Centralized Conferencing Framework
5.2. Conference Control
6. Gateway Considerations
6.1. Gateway Considerations with Textphones
6.2. Gateway Considerations with WebRTC
7. Updates to RFC 4103
8. Congestion Considerations
9. IANA Considerations
9.1. Registration of the "rtt-mixer" SDP Media Attribute
10. Security Considerations
11. References
11.1. Normative References
11.2. Informative References
Acknowledgements
Author's Address
1. Introduction
"RTP Payload for Text Conversation" [RFC4103] specifies the use of
the Real-time Transport Protocol (RTP) [RFC3550] for transmission of
real-time text (often called RTT) and the "text/t140" format. It
also specifies a redundancy format, "text/red", for increased
robustness. The "text/red" format is registered in [RFC4102].
Real-time text is usually provided together with audio and sometimes
with video in conversational sessions.
A requirement related to multiparty sessions from the presentation-
level standard T.140 [T140] for real-time text is as follows:
| The display of text from the members of the conversation should be
| arranged so that the text from each participant is clearly
| readable, and its source and the relative timing of entered text
| is visualized in the display.
Another requirement is that the mixing procedure must not introduce
delays in the text streams that could be perceived as disruptive to
the real-time experience of the receiving users.
The use of real-time text is increasing, and specifically, use in
emergency calls is increasing. Emergency call use requires
multiparty mixing, because it is common that one agent needs to
transfer the call to another specialized agent but is obliged to stay
on the call to at least verify that the transfer was successful.
Mixer implementations for RFC 4103 ("RTP Payload for Text
Conversation") can use traditional RTP functions (RFC 3550) for
mixing and source identification, but the performance of the mixer
when giving turns for the different sources to transmit is limited
when using the default transmission characteristics with redundancy.
The redundancy scheme described in [RFC4103] enables efficient
transmission of earlier transmitted redundant text in packets
together with new text. However, the redundancy header format has no
source indicators for the redundant transmissions. The redundant
parts in a packet must therefore be from the same source as the new
text. The recommended transmission is one new and two redundant
generations of text (T140blocks) in each packet, and the recommended
transmission interval for two-party use is 300 ms.
Real-time text mixers for multiparty sessions need to include the
source with each transmitted group of text from a conference
participant so that the text can be transmitted interleaved with text
groups from different sources at the rate at which they are created.
This enables the text groups to be presented by endpoints in a
suitable grouping with other text from the same source.
The presentation can then be arranged so that text from different
sources can be presented in real time and easily read. At the same
time, it is possible for a reading user to perceive approximately
when the text was created in real time by the different parties. The
transmission and mixing are intended to be done in a general way, so
that presentation can be arranged in a layout decided upon by the
receiving endpoint.
Existing implementations of RFC 4103 in endpoints that do not
implement the updates specified in this document cannot be expected
to properly present real-time text mixed for multiparty-aware
endpoints.
A negotiation mechanism is therefore needed to verify if the parties
(1) are able to handle a common method for multiparty transmissions
and (2) can agree on using that method.
A fallback mixing procedure is also needed for cases when the
negotiation result indicates that a receiving endpoint is not capable
of handling the mixed format. Multiparty-unaware endpoints would
possibly otherwise present all received multiparty mixed text as if
it came from the same source regardless of any accompanying source
indication coded in fields in the packet. Or, they may have other
undesirable ways of acting on the multiparty content. The fallback
method is called the mixing procedure for multiparty-unaware
endpoints. The fallback method is naturally not expected to meet all
performance requirements placed on the mixing procedure for
multiparty-aware endpoints.
This document updates [RFC4103] by introducing an attribute for
declaring support of the RTP-mixer-based multiparty-mixing case and
rules for source indications and interleaving of text from different
sources.
1.1. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
The terms "Source Description" (SDES), "Canonical Name" (CNAME),
"Name" (NAME), "Synchronization Source" (SSRC), "Contributing Source"
(CSRC), "CSRC list", "CSRC count" (CC), "RTP Control Protocol"
(RTCP), and "RTP mixer" are defined in [RFC3550].
"real-time text" (RTT) is text transmitted instantly as it is typed
or created. Recipients can immediately read the message while it is
being written, without waiting.
The term "T140block" is defined in [RFC4103] to contain one or more
T.140 code elements.
"TTY" stands for a textphone type used in North America.
Web Real-Time Communication (WebRTC) is specified by the World Wide
Web Consortium (W3C) and the IETF. See [RFC8825].
"DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension
for use with the Secure Real-time Transport Protocol / Secure Real-
time Transport Control Protocol (SRTP/SRTCP) as specified in
[RFC5764].
The term "multiparty aware" describes an endpoint that (1) receives
real-time text from multiple sources through a common conference
mixer, (2) is able to present the text in real time, separated by
source, and (3) presents the text so that a user can get an
impression of the approximate relative timing of text from different
parties.
The term "multiparty unaware" describes an endpoint that cannot
itself separate text from different sources when the text is received
through a common conference mixer.
1.2. Main Method, Fallback Method, and Considered Alternatives
A number of alternatives were considered when searching for an
efficient and easily implemented multiparty method for real-time
text. This section briefly explains a few of them.
Multiple RTP streams, one per participant:
One RTP stream per source would be sent in the same RTP session
with the "text/red" format. From some points of view, the use of
multiple RTP streams, one for each source, sent in the same RTP
session would be efficient and would use exactly the same packet
format as [RFC4103] and the same payload type. A couple of
relevant scenarios using multiple RTP streams are specified in
"RTP Topologies" [RFC7667]. One possibility of special interest
is the Selective Forwarding Middlebox (SFM) topology specified in
Section 3.7 of [RFC7667], which could enable end-to-end
encryption. In contrast to audio and video, real-time text is
only transmitted when the users actually transmit information.
Thus, an SFM solution would not need to exclude any party from
transmission under normal conditions. In order to allow the mixer
to convey the packets with the payload preserved and encrypted, an
SFM solution would need to act on some specific characteristics of
the "text/red" format. The redundancy headers are part of the
payload, so the receiver would need to just assume that the
payload type number in the redundancy header is for "text/t140".
The characters per second ("cps") parameter would need to act per
stream. The relationship between the SSRC and the source would
need to be conveyed in some specified way, e.g., in the CSRC.
Recovery and loss detection would preferably be based on RTP
sequence number gap detection. Thus, sequence number gaps in the
incoming stream to the mixer would need to be reflected in the
stream to the participant, with no new gaps created by the mixer.
However, the RTP implementation in both mixers and endpoints needs
to support multiple streams in the same RTP session in order to
use this mechanism. To provide the best opportunities for
deployment, it should be possible to upgrade existing endpoint
solutions to be multiparty aware with a reasonable amount of
effort. There is currently a lack of support for multi-stream RTP
in certain implementations. This fact led to only brief mention
of this solution in this document as an option for further study.
RTP-mixer-based method for multiparty-aware endpoints:
The "text/red" format as defined in RFC 4102 and applied in RFC
4103 is sent with the RTP-mixer method indicating the source in
the CSRC field. The "text/red" format with a "text/t140" payload
in a single RTP stream can be sent when text is available from the
call participants instead of at the regular 300 ms intervals.
Transmission of packets with text from different sources can then
be done smoothly while simultaneous transmission occurs as long as
it is not limited by the maximum character rate "cps" value. With
ten participants sending text simultaneously, the switching and
transmission performance is good. With more simultaneously
sending participants and with receivers at default capacity, there
will be a noticeable jerkiness and delay in text presentation.
The more participants who send text simultaneously, the more
jerkiness will occur. Two seconds of jerkiness will be noticeable
and slightly unpleasant, but it corresponds in time to what typing
humans often cause by hesitating or changing position while
typing. A benefit of this method is that no new packet format
needs to be introduced and implemented. Since simultaneous typing
by more than two parties is expected to be very rare -- as
described in Section 1.3 -- this method can be used successfully
with good performance. Recovery of text in the case of packet
loss is based on analysis of timestamps of received redundancy
versus earlier received text. Negotiation is based on a new SDP
media attribute, "rtt-mixer". This method was selected to be the
main method specified in this document.
Multiple sources per packet:
A new "text" media subtype would be specified with up to 15
sources in each packet. The mechanism would make use of the RTP-
mixer model specified in RTP [RFC3550]. The sources would be
indicated in strict order in the CSRC list of the RTP packets.
The CSRC list can have up to 15 members. Therefore, text from up
to 15 sources can be included in each packet. Packets are
normally sent at 300 ms intervals. The mean delay would be 150
ms. A new redundancy packet format would be specified. This
method would result in good performance but would require
standardization and implementation of new releases in the target
technologies; these would take more time than desirable to
complete. It was therefore not selected to be included in this
document.
Mixing for multiparty-unaware endpoints:
The presentation of text from multiple parties is prepared by the
mixer in one single stream. It is desirable to have a method that
does not require any modifications in existing user devices
implementing RFC 4103 for real-time text without explicit support
of multiparty sessions. This is made possible by having the mixer
insert a new line and a text-formatted source label before each
switch of text source in the stream. Switching the source can
only be done in places in the text where it does not disturb the
perception of the contents. Text from only one source at a time
can be presented in real time. The delay will therefore vary. In
calls where parties take turns properly by ending their entries
with a new line, the limitations will have limited influence on
the user experience. When only two parties send text, these two
will see the text in real time with no delay. Although this
method also has other limitations, it is included in this document
as a fallback method.
Real-time text transport in WebRTC:
[RFC8865] specifies how the WebRTC data channel can be used to
transport real-time text. That specification contains a section
briefly describing its use in multiparty sessions. The focus of
this document is RTP transport. Therefore, even if the WebRTC
transport provides good multiparty performance, it is only
mentioned in this document in relation to providing gateways with
multiparty capabilities between RTP and WebRTC technologies.
1.3. Intended Application
The method for multiparty real-time text specified in this document
is primarily intended for use in transmissions between mixers and
endpoints in centralized mixing configurations. It is also
applicable between mixers. An often-mentioned application is for
emergency service calls with real-time text and voice, where a call
taker wants to make an attended handover of a call to another agent
and stay on the call to observe the session. Multimedia conference
sessions with support for participants to contribute with text is
another example. Conferences with central support for speech-to-text
conversion represent yet another example.
In all these applications, normally only one participant at a time
will send long text comments. In some cases, one other participant
will occasionally contribute with a longer comment simultaneously.
That may also happen in some rare cases when text is translated to
text in another language in a conference. Apart from these cases,
other participants are only expected to contribute with very brief
comments while others are sending text.
Users expect the text they send to be presented in real time in a
readable way to the other participants even if they send
simultaneously with other users and even when they make brief edit
operations of their text by backspacing and correcting their text.
Text is supposed to be human generated, by some means of text input,
such as typing on a keyboard or using speech-to-text technology.
Occasional small cut-and-paste operations may appear even if that is
not the initial purpose of real-time text.
The real-time characteristics of real-time text are essential for the
participants to be able to contribute to a conversation. If the text
is delayed too much between the typing of a character and its
presentation, then, in some conference situations, the opportunity to
comment will be gone and someone else will grab the turn. A delay of
more than one second in such situations is an obstacle to good
conversation.
2. Overview of the Two Specified Solutions and Selection of Method
This section contains a brief introduction of the two methods
specified in this document.
2.1. The RTP-Mixer-Based Solution for Multiparty-Aware Endpoints
This method specifies the negotiated use of the formats described in
RFC 4103, for multiparty transmissions in a single RTP stream. The
main purpose of this document is to specify a method for true
multiparty real-time text mixing for multiparty-aware endpoints that
can be widely deployed. The RTP-mixer-based method makes use of the
current format for real-time text as provided in [RFC4103]. This
method updates RFC 4103 by clarifying one way to use it in the
multiparty situation. That is done by completing a negotiation for
this kind of multiparty capability and by interleaving packets from
different sources. The source is indicated in the CSRC element in
the RTP packets. Specific considerations are made regarding the
ability to recover text after packet loss.
The detailed procedures for the RTP-mixer-based multiparty-aware case
are specified in Section 3.
Please refer to [RFC4103] when reading this document.
2.2. Mixing for Multiparty-Unaware Endpoints
This document also specifies a method to be used in cases when the
endpoint participating in a multiparty call does not itself implement
any solution or does not implement the same solution as the mixer.
This method requires the mixer to insert text dividers and readable
labels and only send text from one source at a time until a suitable
point appears for changing the source. This solution is a fallback
method with functional limitations. It operates at the presentation
level.
A mixer SHOULD by default format and transmit text to a call
participant so that the text is suitable for presentation on a
multiparty-unaware endpoint that has not negotiated any method for
true multiparty real-time text handling but has negotiated a "text/
red" or "text/t140" format in a session. This SHOULD be done if
nothing else is specified for the application, in order to maintain
interoperability. Section 4.2 specifies how this mixing is done.
2.3. Offer/Answer Considerations
"RTP Payload for Text Conversation" [RFC4103] specifies the use of
RTP [RFC3550] and a redundancy format ("text/red", as defined in
[RFC4102]) for increased robustness of real-time text transmission.
This document updates [RFC4103] by introducing a capability
negotiation for handling multiparty real-time text, a way to indicate
the source of transmitted text, and rules for efficient timing of the
transmissions interleaved from different sources.
The capability negotiation for the RTP-mixer-based multiparty method
is based on the use of the SDP media attribute "rtt-mixer".
The syntax is as follows:
a=rtt-mixer
If in the future any other method for RTP-based multiparty real-time
text is specified by additional work, it is assumed that it will be
recognized by some specific SDP feature exchange.
2.3.1. Initial Offer
A party that intends to set up a session and is willing to use the
RTP-mixer-based method provided in this specification for sending,
receiving, or both sending and receiving real-time text SHALL include
the "rtt-mixer" SDP attribute in the corresponding "text" media
section in the initial offer.
The party MAY indicate its capability regarding both the RTP-mixer-
based method provided in this specification and other methods.
When the offerer has sent the offer, which includes the "rtt-mixer"
attribute, it MUST be prepared to receive and handle real-time text
formatted according to both the method for multiparty-aware parties
specified in Section 3 and two-party formatted real-time text.
2.3.2. Answering the Offer
A party that receives an offer containing the "rtt-mixer" SDP
attribute and is willing to use the RTP-mixer-based method provided
in this specification for sending, receiving, or both sending and
receiving real-time text SHALL include the "rtt-mixer" SDP attribute
in the corresponding "text" media section in the answer.
If the offer did not contain the "rtt-mixer" attribute, the answer
MUST NOT contain the "rtt-mixer" attribute.
Even when the "rtt-mixer" attribute is successfully negotiated, the
parties MAY send and receive two-party coded real-time text.
An answer MUST NOT include acceptance of more than one method for
multiparty real-time text in the same RTP session.
When the answer, which includes acceptance, is transmitted, the
answerer MUST be prepared to act on received text in the negotiated
session according to the method for multiparty-aware parties
specified in Section 3. Reception of text for a two-party session
SHALL also be supported.
2.3.3. Offerer Processing the Answer
When the answer is processed by the offerer, the offerer MUST follow
the requirements listed in Section 2.4.
2.3.4. Modifying a Session
A session MAY be modified at any time by any party offering a
modified SDP with or without the "rtt-mixer" SDP attribute expressing
a desired change in the support of multiparty real-time text.
If the modified offer adds the indication of support for multiparty
real-time text by including the "rtt-mixer" SDP attribute, the
procedures specified in the previous subsections SHALL be applied.
If the modified offer deletes the indication of support for
multiparty real-time text by excluding the "rtt-mixer" SDP attribute,
the answer MUST NOT contain the "rtt-mixer" attribute. After
processing this SDP exchange, the parties MUST NOT send real-time
text formatted for multiparty-aware parties according to this
specification.
2.4. Actions Depending on Capability Negotiation Result
A transmitting party SHALL send text according to the RTP-mixer-based
multiparty method only when the negotiation for that method was
successful and when it conveys text for another source. In all other
cases, the packets SHALL be populated and interpreted as for a two-
party session.
A party that has negotiated the "rtt-mixer" SDP media attribute and
acts as an RTP mixer sending multiparty text MUST (1) populate the
CSRC list and (2) format the packets according to Section 3.
A party that has negotiated the "rtt-mixer" SDP media attribute MUST
interpret the contents of the CC field, the CSRC list, and the
packets according to Section 3 in received RTP packets in the
corresponding RTP stream.
A party that has not successfully completed the negotiation of the
"rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved
from different sources in the same RTP stream, as specified in
Section 3. If the party is a mixer and did declare the "rtt-mixer"
SDP media attribute, it SHOULD perform the procedure for multiparty-
unaware endpoints. If the party is not a mixer, it SHOULD transmit
as in a two-party session according to [RFC4103].
3. Details for the RTP-Mixer-Based Mixing Method for Multiparty-Aware
Endpoints
3.1. Use of Fields in the RTP Packets
The CC field SHALL show the number of members in the CSRC list, which
SHALL be one (1) in transmissions from a mixer when conveying text
from other sources in a multiparty session, and otherwise 0.
When text is conveyed by a mixer during a multiparty session, a CSRC
list SHALL be included in the packet. The single member in the CSRC
list SHALL contain the SSRC of the source of the T140blocks in the
packet.
When redundancy is used, the RECOMMENDED level of redundancy is to
use one primary and two redundant generations of T140blocks. In some
cases, a primary or redundant T140block is empty but is still
represented by a member in the redundancy header.
In other respects, the contents of the RTP packets will be as
specified in [RFC4103].
3.2. Initial Transmission of a BOM Character
As soon as a participant is known to participate in a session with
another entity and is available for text reception, a Unicode byte
order mark (BOM) character SHALL be sent to it by the other entity
according to the procedures in this section. This is useful in many
configurations for opening ports and firewalls and for setting up the
connection between the application and the network. If the
transmitter is a mixer, then the source of this character SHALL be
indicated to be the mixer itself.
Note that the BOM character SHALL be transmitted with the same
redundancy procedures as any other text.
3.3. Keep-Alive
After that, the transmitter SHALL send keep-alive traffic to the
receiver(s) at regular intervals when no other traffic has occurred
during that interval, if that is decided upon for the actual
connection. It is RECOMMENDED to use the keep-alive solution
provided in [RFC6263]. The consent check [RFC7675] is a possible
alternative if it is used anyway for other reasons.
3.4. Transmission Interval
A "text/red" or "text/t140" transmitter in a mixer SHALL send packets
distributed over time as long as there is something (new or redundant
T140blocks) to transmit. The maximum transmission interval between
text transmissions from the same source SHALL then be 330 ms, when no
other limitations cause a longer interval to be temporarily used. It
is RECOMMENDED to send the next packet to a receiver as soon as new
text to that receiver is available, as long as the mean character
rate of new text to the receiver calculated over the last 10 one-
second intervals does not exceed the "cps" value of the receiver.
The intention is to keep the latency low and network load limited
while keeping good protection against text loss in bursty packet loss
conditions. The main purpose of the 330 ms interval is for the
timing of redundant transmissions, when no new text from the same
source is available.
The value of 330 ms is used, because many sources of text will
transmit new text at 300 ms intervals during periods of continuous
user typing, and then reception in the mixer of such new text will
cause a combined transmission of the new text and the unsent
redundancy from the previous transmission. Only when the user stops
typing will the 330 ms interval be applied to send the redundancy.
If the characters per second ("cps") value is reached, a longer
transmission interval SHALL be applied for text from all sources as
specified in [RFC4103] and only as much of the text queued for
transmission SHALL be sent at the end of each transmission interval
as can be allowed without exceeding the "cps" value. Division of
text for partial transmission MUST then be made at T140block borders.
When the transmission rate falls below the "cps" value again, the
transmission intervals SHALL be reset to 330 ms and transmission of
new text SHALL again be made as soon as new text is available.
| NOTE: Extending the transmission intervals during periods of
| high load does not change the number of characters to be
| conveyed. It just evens out the load over time and reduces the
| number of packets per second. With human-created
| conversational text, the sending user will eventually take a
| pause, letting transmission catch up.
See also Section 8.
For a transmitter not acting as a mixer, the transmission interval
principles provided in [RFC4103] apply, and the normal transmission
interval SHALL be 300 ms.
3.5. Only One Source per Packet
New text and redundant copies of earlier text from one source SHALL
be transmitted in the same packet if available for transmission at
the same time. Text from different sources MUST NOT be transmitted
in the same packet.
3.6. Do Not Send Received Text to the Originating Source
Text received by a mixer from a participant SHOULD NOT be included in
transmissions from the mixer to that participant, because for text
that is produced locally, the normal behavior of the endpoint is to
present such text directly when it is produced.
3.7. Clean Incoming Text
A mixer SHALL handle reception, recovery from packet loss, deletion
of superfluous redundancy, marking of possible text loss, and
deletion of BOM characters from each participant before queueing
received text for transmission to receiving participants as specified
in [RFC4103] for single-party sources and Section 3.16 for multiparty
sources (chained mixers).
3.8. Principles of Redundant Transmission
A transmitting party using redundancy SHALL send redundant
repetitions of T140blocks already transmitted in earlier packets.
The number of redundant generations of T140blocks to include in
transmitted packets SHALL be deduced from the SDP negotiation. It
SHALL be set to the minimum of the number declared by the two parties
negotiating a connection. It is RECOMMENDED to declare and transmit
one original and two redundant generations of the T140blocks, because
this provides good protection against text loss in the case of packet
loss and also provides low overhead.
3.9. Text Placement in Packets
The mixer SHALL compose and transmit an RTP packet to a receiver when
one or more of the following conditions have occurred:
* The transmission interval is the normal 330 ms (no matter whether
the transmission interval has passed or not), and there is newly
received unsent text available for transmission to that receiver.
* The current transmission interval has passed and is longer than
the normal 330 ms, and there is newly received unsent text
available for transmission to that receiver.
* The current transmission interval (normally 330 ms) has passed
since already-transmitted text was queued for transmission as
redundant text.
The principles provided in [RFC4103] apply for populating the header,
the redundancy header, and the data in the packet with specific
information, as detailed here and in the following sections.
At the time of transmission, the mixer SHALL populate the RTP packet
with all T140blocks queued for transmission originating from the
source selected for transmission as long as this is not in conflict
with the allowed number of characters per second ("cps") or the
maximum packet size. In this way, the latency of the latest received
text is kept low even in moments of simultaneous transmission from
many sources.
Redundant text SHALL also be included, and the assessment of how much
new text can be included within the maximum packet size MUST take
into account that the redundancy has priority to be transmitted in
its entirety. See Section 3.4.
The SSRC of the source SHALL be placed as the only member in the CSRC
list.
| Note: The CSRC list in an RTP packet only includes the
| participant whose text is included in text blocks. It is not
| the same as the total list of participants in a conference.
| With audio and video media, the CSRC list would often contain
| all participants who are not muted, whereas text participants
| that don't type are completely silent and thus are not
| represented in RTP packet CSRC lists.
3.10. Empty T140blocks
If no unsent T140blocks were available for a source at the time of
populating a packet but already-transmitted T140blocks are available
that have not yet been sent the full intended number of redundant
transmissions, then the primary area in the packet is composed of an
empty T140block and included (without taking up any length) in the
packet for transmission. The corresponding SSRC SHALL be placed as
usual in its place in the CSRC list.
The first packet in the session, the first after a source switch, and
the first after a pause SHALL be populated with the available
T140blocks for the source selected to be sent as the primary, and
empty T140blocks for the agreed-upon number of redundancy
generations.
3.11. Creation of the Redundancy
The primary T140block from a source in the latest transmitted packet
is saved for populating the first redundant T140block for that source
in the next transmission of text from that source. The first
redundant T140block for that source from the latest transmission is
saved for populating the second redundant T140block in the next
transmission of text from that source.
Usually, this is the level of redundancy used. If a higher level of
redundancy is negotiated, then the procedure SHALL be continued until
all available redundant levels of T140blocks are placed in the
packet. If a receiver has negotiated a lower number of "text/red"
generations, then that level SHALL be the maximum used by the
transmitter.
The T140blocks saved for transmission as redundant data are assigned
a planned transmission time of 330 ms after the current time but
SHOULD be transmitted earlier if new text for the same source gets
selected for transmission before that time.
3.12. Timer Offset Fields
The timestamp offset values SHALL be inserted in the redundancy
header, with the time offset from the RTP timestamp in the packet
when the corresponding T140block was sent as the primary.
The timestamp offsets are expressed in the same clock tick units as
the RTP timestamp.
The timestamp offset values for empty T140blocks have no relevance
but SHOULD be assigned realistic values.
3.13. Other RTP Header Fields
The number of members in the CSRC list (0 or 1) SHALL be placed in
the CC header field. Only mixers place value 1 in the CC field. A
value of 0 indicates that the source is the transmitting device
itself and that the source is indicated by the SSRC field. This
value is used by endpoints and also by mixers sending self-sourced
data.
The current time SHALL be inserted in the timestamp.
The SSRC header field SHALL contain the SSRC of the RTP session where
the packet will be transmitted.
The M-bit SHALL be handled as specified in [RFC4103].
3.14. Pause in Transmission
When there is no new T140block to transmit and no redundant T140block
that has not been retransmitted the intended number of times from any
source, the transmission process SHALL be stopped until either new
T140blocks arrive or a keep-alive method calls for transmission of
keep-alive packets.
3.15. RTCP Considerations
A mixer SHALL send RTCP reports with SDES, CNAME, and NAME
information about the sources in the multiparty call. This makes it
possible for participants to compose a suitable label for text from
each source.
Privacy considerations SHALL be taken when composing these fields.
They contain name and address information that may be considered
sensitive if the information is transmitted in its entirety, e.g., to
unauthenticated participants.
3.16. Reception of Multiparty Contents
The "text/red" receiver included in an endpoint with presentation
functions will receive RTP packets in the single stream from the
mixer and SHALL distribute the T140blocks for presentation in
presentation areas for each source. Other receiver roles, such as
gateways or chained mixers, are also feasible. Whether the stream
will only be forwarded or will be distributed based on the different
sources must be taken into consideration.
3.16.1. Acting on the Source of the Packet Contents
If the CC field value of a received packet is 1, it indicates that
the text is conveyed from a source indicated in the single member in
the CSRC list, and the receiver MUST act on the source according to
its role. If the CC value is 0, the source is indicated in the SSRC
field.
3.16.2. Detection and Indication of Possible Text Loss
The receiver SHALL monitor the RTP sequence numbers of the received
packets for gaps and for packets received out of order. If a
sequence number gap appears and still exists after some defined short
time for jitter and reordering resolution, the packets in the gap
SHALL be regarded as lost.
If it is known that only one source is active in the RTP session,
then it is likely that a gap equal to or larger than the agreed-upon
number of redundancy generations (including the primary) causes text
loss. In that case, the receiver SHALL create a T140block with a
marker for possible text loss [T140ad1], associate it with the
source, and insert it in the reception buffer for that source.
If it is known that more than one source is active in the RTP
session, then it is not possible in general to evaluate if text was
lost when packets were lost. With two active sources and the
recommended number of redundancy generations (one original and two
redundant), it can take a gap of five consecutive lost packets before
any text may be lost, but text loss can also appear if three non-
consecutive packets are lost when they contained consecutive data
from the same source. A simple method for deciding when there is a
risk of resulting text loss is to evaluate if three or more packets
were lost within one second. If this simple method is used, then a
T140block SHOULD be created with a marker for possible text loss
[T140ad1] and associated with the SSRC of the RTP session as a
general input from the mixer.
Implementations MAY apply more refined methods for more reliable
detection of whether text was lost or not. Any refined method SHOULD
prefer marking possible loss rather than not marking when it is
uncertain if there was loss.
3.16.3. Extracting Text and Handling Recovery
When applying the following procedures, the effects of possible
timestamp wraparound and the RTP session possibly changing the SSRC
MUST be considered.
When a packet is received in an RTP session using the packetization
for multiparty-aware endpoints, its T140blocks SHALL be extracted as
described below.
The source SHALL be extracted from the CSRC list if available, and
otherwise from the SSRC.
If the received packet is the first packet received from the source,
then all T140blocks in the packet SHALL be retrieved and assigned to
a receive buffer for that source, beginning with the oldest available
redundant generation, continuing with the younger redundant
generations in age order, and finally ending with the primary.
| Note: The normal case is that in the first packet, only the
| primary data has contents. The redundant data has contents in
| the first received packet from a source only after initial
| packet loss.
If the packet is not the first packet from a source, then if
redundant data is available, the process SHALL start with the oldest
generation. The timestamp of that redundant data SHALL be created by
subtracting its timestamp offset from the RTP timestamp. If the
resulting timestamp is later than the latest retrieved data from the
same source, then the redundant data SHALL be retrieved and appended
to the receive buffer. The process SHALL be continued in the same
way for all younger generations of redundant data. After that, the
timestamp of the packet SHALL be compared with the timestamp of the
latest retrieved data from the same source and if it is later, then
the primary data SHALL be retrieved from the packet and appended to
the receive buffer for the source.
3.16.4. Delete BOM
The Unicode BOM character is used as a start indication and is
sometimes used as a filler or keep-alive by transmission
implementations. Any BOM characters SHALL be deleted after
extraction from received packets.
3.17. Performance Considerations
This solution has good performance with low text delays, as long as
the mean number of characters per second sent during any 10-second
interval from a number of simultaneously sending participants to a
receiving participant does not reach the "cps" value. At higher
numbers of sent characters per second, a jerkiness is visible in the
presentation of text. The solution is therefore suitable for
emergency service use, relay service use, and small or well-managed
larger multimedia conferences. In large unmanaged conferences with a
high number of participants only, on very rare occasions, situations
might arise where many participants happen to send text
simultaneously. In such circumstances, the result may be
unpleasantly jerky presentation of text from each sending
participant. It should be noted that it is only the number of users
sending text within the same moment that causes jerkiness, not the
total number of users with real-time text capability.
3.18. Security for Session Control and Media
Security mechanisms to provide confidentiality, integrity protection,
and peer authentication SHOULD be applied when possible regarding the
capabilities of the participating devices by using the Session
Initiation Protocol (SIP) over TLS by default according to
Section 3.1.3 of [RFC5630] on the session control level and by
default using DTLS-SRTP [RFC5764] at the media level. In
applications where legacy endpoints without security are allowed, a
negotiation SHOULD be performed to decide if encryption at the media
level will be applied. If no other security solution is mandated for
the application, then the Opportunistic Secure Real-time Transport
Protocol (OSRTP) [RFC8643] is a suitable method to be applied to
negotiate SRTP media security with DTLS. For simplicity, most SDP
examples below are expressed without the security additions. The
principles (but not all details) for applying DTLS-SRTP security
[RFC5764] are shown in a couple of the following examples.
Further general security considerations are covered in Section 10.
End-to-end encryption would require further work and could be based
on WebRTC as specified in Section 1.2 or on double encryption as
specified in [RFC8723].
3.19. SDP Offer/Answer Examples
This section shows some examples of SDP for session negotiation of
the real-time text media in SIP sessions. Audio is usually provided
in the same session, and sometimes also video. The examples only
show the part of importance for the real-time text media. The
examples relate to the single RTP stream mixing for multiparty-aware
endpoints and for multiparty-unaware endpoints.
| Note: Multiparty real-time text MAY also be provided through
| other methods, e.g., by a Selective Forwarding Middlebox (SFM).
| In that case, the SDP of the offer will include something
| specific for that method, e.g., an SDP attribute or another
| media format. An answer selecting the use of that method would
| accept it via a corresponding acknowledgement included in the
| SDP. The offer may also contain the "rtt-mixer" SDP media
| attribute for the main real-time text media when the offerer
| has this capability for both multiparty methods, while an
| answer, choosing to use SFM, will not include the "rtt-mixer"
| SDP media attribute.
Offer example for the "text/red" format, multiparty support, and
capability for 90 characters per second:
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=fmtp:98 cps=90
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Answer example from a multiparty-aware device:
m=text 14000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=fmtp:98 cps=90
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Offer example for the "text/red" format, including multiparty and
security:
a=fingerprint: (fingerprint1)
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
The "fingerprint" is sufficient to offer DTLS-SRTP, with the media
line still indicating RTP/AVP.
| Note: For brevity, the entire value of the SDP "fingerprint"
| attribute is not shown in this and the following example.
Answer example from a multiparty-aware device with security:
a=fingerprint: (fingerprint2)
m=text 16000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
With the "fingerprint", the device acknowledges the use of DTLS-SRTP.
Answer example from a multiparty-unaware device that also does not
support security:
m=text 12000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
3.20. Packet Sequence Example from Interleaved Transmission
This example shows a symbolic flow of packets from a mixer, including
loss and recovery. The sequence includes interleaved transmission of
text from two real-time text sources: A and B. P indicates primary
data. R1 is the first redundant generation of data, and R2 is the
second redundant generation of data. A1, B1, A2, etc. are text
chunks (T140blocks) received from the respective sources and sent on
to the receiver by the mixer. X indicates a dropped packet between
the mixer and a receiver. The session is assumed to use the original
and two redundant generations of real-time text.
|-----------------------|
|Seq no 101, Time=20400 |
|CC=1 |
|CSRC list A |
|R2: A1, Offset=600 |
|R1: A2, Offset=300 |
|P: A3 |
|-----------------------|
Assuming that earlier packets (with text A1 and A2) were received in
sequence, text A3 is received from packet 101 and assigned to
reception buffer A. The mixer is now assumed to have received
initial text from source B 100 ms after packet 101 and will send that
text. Transmission of A2 and A3 as redundancy is planned for 330 ms
after packet 101 if no new text from A is ready to be sent before
that.
|-----------------------|
|Seq no 102, Time=20500 |
|CC=1 |
|CSRC list B |
|R2 Empty, Offset=600 |
|R1: Empty, Offset=300 |
|P: B1 |
|-----------------------|
Packet 102 is received.
B1 is retrieved from this packet. Redundant transmission of B1 is
planned 330 ms after packet 102.
X------------------------|
X Seq no 103, Timer=20730|
X CC=1 |
X CSRC list A |
X R2: A2, Offset=630 |
X R1: A3, Offset=330 |
X P: Empty |
X------------------------|
Packet 103 is assumed to be lost due to network problems.
It contains redundancy for A. Sending A3 as second-level
redundancy is planned for 330 ms after packet 103.
X------------------------|
X Seq no 104, Timer=20800|
X CC=1 |
X CSRC list B |
X R2: Empty, Offset=600 |
X R1: B1, Offset=300 |
X P: B2 |
X------------------------|
Packet 104 contains text from B, including new B2 and redundant
B1. It is assumed dropped due to network problems.
The mixer has A3 redundancy to send, but no new text appears from
A, and therefore the redundancy is sent 330 ms after the previous
packet with text from A.
|------------------------|
| Seq no 105, Timer=21060|
| CC=1 |
| CSRC list A |
| R2: A3, Offset=660 |
| R1: Empty, Offset=330 |
| P: Empty |
|------------------------|
Packet 105 is received.
A gap for lost packets 103 and 104 is detected. Assume that no
other loss was detected during the last second. It can then be
concluded that nothing was totally lost.
R2 is checked. Its original time was 21060-660=20400. A packet
with text from A was received with that timestamp, so nothing
needs to be recovered.
B1 and B2 still need to be transmitted as redundancy. This is
planned 330 ms after packet 104. That would be at 21130.
|-----------------------|
|Seq no 106, Timer=21130|
|CC=1 |
|CSRC list B |
| R2: B1, Offset=630 |
| R1: B2, Offset=330 |
| P: Empty |
|-----------------------|
Packet 106 is received.
The second-level redundancy in packet 106 is B1 and has a
timestamp offset of 630 ms. The timestamp of packet 106 minus 630
is 20500, which is the timestamp of packet 102 that was received.
So, B1 does not need to be retrieved. The first-level redundancy
in packet 106 has an offset of 330. The timestamp of packet 106
minus 330 is 20800. That is later than the latest received packet
with source B. Therefore, B2 is retrieved and assigned to the
input buffer for source B. No primary is available in packet 106.
After this sequence, A3, B1, and B2 have been received. In this
case, no text was lost.
3.21. Maximum Character Rate "cps" Setting
The default maximum rate of reception of "text/t140" real-time text,
as specified in [RFC4103], is 30 characters per second. The actual
rate is calculated without regard to any redundant text transmission
and is, in the multiparty case, evaluated for all sources
contributing to transmission to a receiver. The value MAY be
modified in the "cps" parameter of the "fmtp" attribute for the
"text/t140" format of the "text" media section.
A mixer combining real-time text from a number of sources may
occasionally have a higher combined flow of text coming from the
sources. Endpoints SHOULD therefore include a suitable higher value
for the "cps" parameter, corresponding to its real reception
capability. The default "cps" value 30 can be assumed to be
sufficient for small meetings and well-managed larger conferences
with users only making manual text entry. A "cps" value of 90 can be
assumed to be sufficient even for large unmanaged conferences and for
cases when speech-to-text technologies are used for text entry. This
is also a reachable performance for receivers in modern technologies,
and 90 is therefore the RECOMMENDED "cps" value. See [RFC4103] for
the format and use of the "cps" parameter. The same rules apply for
the multiparty case.
4. Presentation-Level Considerations
"Protocol for multimedia application text conversation" [T140]
provides the presentation-level requirements for RTP transport as
described in [RFC4103]. Functions for erasure and other formatting
functions are specified in [T140], which has the following general
statement for the presentation:
| The display of text from the members of the conversation should be
| arranged so that the text from each participant is clearly
| readable, and its source and the relative timing of entered text
| is visualized in the display. Mechanisms for looking back in the
| contents from the current session should be provided. The text
| should be displayed as soon as it is received.
Strict application of [T140] is essential for the interoperability of
real-time text implementations and to fulfill the intention that the
session participants have the same information conveyed in the text
contents of the conversation without necessarily having the exact
same layout of the conversation.
[T140] specifies a set of presentation control codes (Section 4.2.4)
to include in the stream. Some of them are optional.
Implementations MUST ignore optional control codes that they do not
support.
There is no strict "message" concept in real-time text. The Unicode
Line Separator character SHALL be used as a separator allowing a part
of received text to be grouped in a presentation. The character
combination "CRLF" may be used by other implementations as a
replacement for the Line Separator. The "CRLF" combination SHALL be
erased by just one erasing action, the same as the Line Separator.
Presentation functions are allowed to group text for presentation in
smaller groups than the Line Separators imply and present such groups
with a source indication together with text groups from other sources
(see the following presentation examples). Erasure has no specific
limit by any delimiter in the text stream.
4.1. Presentation by Multiparty-Aware Endpoints
A multiparty-aware receiving party presenting real-time text MUST
separate text from different sources and present them in separate
presentation fields. The receiving party MAY separate the
presentation of parts of text from a source in readable groups based
on criteria other than a Line Separator and merge these groups in the
presentation area when it benefits the user to most easily find and
read text from the different participants. The criteria MAY, for
example, be a received comma, a full stop, some other type of phrase
delimiter, or a long pause.
When text is received from multiple original sources, the
presentation SHALL provide a view where text is added in multiple
presentation fields.
If the presentation presents text from different sources in one
common area, the presenting endpoint SHOULD insert text from the
local user, where the text ends at suitable points and is merged
properly with received text to indicate the relative timing for when
the text groups were completed. In this presentation mode, the
receiving endpoint SHALL present the source of the different groups
of text. This presentation style is called the "chat" style here and
provides the possibility of following text arriving from multiple
parties and the approximate relative time that text is received as
related to text from the local user.
A view of a three-party real-time text call in chat style 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 |-|
|______________________________________________|v|
| <Eve-typing> But I need to be back to |^|
| the hotel by 11 because I need |-|
| | |
| <Bob-typing> I wou |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 1: Example of a Three-Party Real-Time Text Call Presented
in Chat Style Seen at Participant Alice's Endpoint
Presentation styles other than the chat style MAY be arranged.
Figure 2 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: An Example of a Coordinated Column View of a
Three-Party Session with Entries Ordered Vertically in
Approximate Time Order
4.2. Multiparty Mixing for Multiparty-Unaware Endpoints
When the mixer has indicated multiparty real-time text capability in
an SDP negotiation but the multiparty capability negotiation fails
with an endpoint, the agreed-upon "text/red" or "text/t140" format
SHALL be used and the mixer SHOULD compose a best-effort presentation
of multiparty real-time text in one stream intended to be presented
by an endpoint with no multiparty awareness, when that is desired in
the actual implementation. The following specifies a procedure that
MAY be applied in that situation.
This presentation format has functional limitations and SHOULD be
used only to enable participation in multiparty calls by legacy
deployed endpoints implementing only RFC 4103 without any multiparty
extensions specified in this document.
The principles and procedures below do not specify any new protocol
elements. They are instead composed of information provided in
[T140] and an ambition to provide a best-effort presentation on an
endpoint that has functions originally intended only for two-party
calls.
The mixer performing the mixing for multiparty-unaware endpoints
SHALL compose a simulated, limited multiparty real-time text view
suitable for presentation in one presentation area. The mixer SHALL
group text in suitable groups and prepare them for presentation by
inserting a Line Separator between them if the transmitted text did
not already end with a new line (Line Separator or CRLF). A
presentable label SHALL be composed and sent for the source initially
in the session and after each source switch. With this procedure,
the time for switching from transmission of text from one source to
transmission of text from another source depends on the actions of
the users. In order to expedite source switching, a user can, for
example, end its turn with a new line.
4.2.1. Actions by the Mixer at Reception from the Call Participants
When text is received by the mixer from the different participants,
the mixer SHALL recover text from redundancy if any packets are lost.
The marker for lost text [T140ad1] SHALL be inserted in the stream if
unrecoverable loss appears. Any Unicode BOM characters, possibly
used for keep-alives, SHALL be deleted. The time of creation of text
(retrieved from the RTP timestamp) SHALL be stored together with the
received text from each source in queues for transmission to the
recipients in order to be able to evaluate text loss.
4.2.2. Actions by the Mixer for Transmission to the Recipients
The following procedure SHALL be applied for each multiparty-unaware
recipient of multiparty text from the mixer.
The text for transmission SHALL be formatted by the mixer for each
receiving user for presentation in one single presentation area.
Text received from a participant SHOULD NOT be included in
transmissions to that participant, because it is usually presented
locally at transmission time. When there is text available for
transmission from the mixer to a receiving party from more than one
participant, the mixer SHALL switch between transmission of text from
the different sources at suitable points in the transmitted stream.
When switching the source, the mixer SHALL insert a Line Separator if
the already-transmitted text did not end with a new line (Line
Separator or CRLF). A label SHALL be composed of information in the
CNAME and NAME fields in RTCP reports from the participant to have
its text transmitted, or from other session information for that
user. The label SHALL be delimited by suitable characters (e.g.,
"[ ]") and transmitted. The CSRC SHALL indicate the selected source.
Then, text from that selected participant SHALL be transmitted until
a new suitable point for switching the source is reached.
Information available to the mixer for composing the label may
contain sensitive personal information that SHOULD NOT be revealed in
sessions not securely authenticated and confidentiality protected.
Privacy considerations regarding how much personal information is
included in the label SHOULD therefore be taken when composing the
label.
Seeking a suitable point for switching the source SHALL be done when
there is older text waiting for transmission from any party than the
age of the last transmitted text. Suitable points for switching are:
* A completed phrase ending with a comma.
* A completed sentence.
* A new line (Line Separator or CRLF).
* A long pause (e.g., > 10 seconds) in received text from the
currently transmitted source.
* If text from one participant has been transmitted with text from
other sources waiting for transmission for a long time (e.g., > 1
minute) and none of the other suitable points for switching has
occurred, a source switch MAY be forced by the mixer at the next
word delimiter, and also even if a word delimiter does not occur
within some period of time (e.g., 15 seconds) after the scan for a
word delimiter started.
When switching the source, the source that has the oldest text in
queue SHALL be selected to be transmitted. A character display count
SHALL be maintained for the currently transmitted source, starting at
zero after the label is transmitted for the currently transmitted
source.
The status SHALL be maintained for the latest control code for Select
Graphic Rendition (SGR) from each source. If there is an SGR code
stored as the status for the current source before the source switch
is done, a reset of SGR SHALL be sent by the sequence SGR 0 [U+009B
U+0000 U+006D] after the new line and before the new label during a
source switch. See Section 4.2.4 for an explanation. This
transmission does not influence the display count.
If there is an SGR code stored for the new source after the source
switch, that SGR code SHALL be transmitted to the recipient before
the label. This transmission does not influence the display count.
4.2.3. Actions on Transmission of Text
Text from a source sent to the recipient SHALL increase the display
count by one per transmitted character.
4.2.4. Actions on Transmission of Control Codes
The following control codes, as specified by T.140 [T140], require
specific actions. They SHALL cause specific considerations in the
mixer. Note that the codes presented here are expressed in UTF-16,
while transmission is made in the UTF-8 encoding of these codes.
BEL (U+0007): Bell. Alert in session. Provides for alerting during
an active session. The display count SHALL NOT be altered.
NEW LINE (U+2028): Line Separator. Check and perform a source
switch if appropriate. Increase the display count by 1.
CR LF (U+000D U+000A): A supported, but not preferred, way of
requesting a new line. Check and perform a source switch if
appropriate. Increase the display count by 1.
INT (ESC U+0061): Interrupt (used to initiate the mode negotiation
procedure). The display count SHALL NOT be altered.
SGR (U+009B Ps U+006D): Select Graphic Rendition. Ps represents the
rendition parameters specified in [ISO6429]. (For freely
available equivalent information, please see [ECMA-48].) The
display count SHALL NOT be altered. The SGR code SHOULD be stored
for the current source.
SOS (U+0098): Start of String. Used as a general protocol element
introducer, followed by a maximum 256-byte string and the ST. The
display count SHALL NOT be altered.
ST (U+009C): String Terminator. End of SOS string. The display
count SHALL NOT be altered.
ESC (U+001B): Escape. Used in control strings. The display count
SHALL NOT be altered for the complete escape code.
Byte order mark (BOM) (U+FEFF): "Zero width no-break space". Used
for synchronization and keep-alive. It SHALL be deleted from
incoming streams. It SHALL also be sent first after session
establishment to the recipient. The display count SHALL NOT be
altered.
Missing text mark (U+FFFD): "Replacement character". Represented as
a question mark in a rhombus, or, if that is not feasible,
replaced by an apostrophe ('). It marks the place in the stream
of possible text loss. This mark SHALL be inserted by the
reception procedure in the case of unrecoverable loss of packets.
The display count SHALL be increased by one when sent as for any
other character.
SGR: If a control code for SGR other than a reset of the graphic
rendition (SGR 0) is sent to a recipient, that control code SHALL
also be stored as the status for the source in the storage for SGR
status. If a reset graphic rendition (SGR 0) originating from a
source is sent, then the SGR status storage for that source SHALL
be cleared. The display count SHALL NOT be increased.
BS (U+0008): "Back Space". Intended to erase the last entered
character by a source. Erasure by backspace cannot always be
performed as the erasing party intended. If an erasing action
erases all text up to the end of the leading label after a source
switch, then the mixer MUST NOT transmit more backspaces.
Instead, it is RECOMMENDED that a letter "X" be inserted in the
text stream for each backspace as an indication of the intent to
erase more. A new line is usually coded by a Line Separator, but
the character combination "CRLF" MAY be used instead. Erasure of
a new line is, in both cases, done by just one erasing action
(backspace). If the display count has a positive value, it SHALL
be decreased by one when the BS is sent. If the display count is
at zero, it SHALL NOT be altered.
4.2.5. Packet Transmission
A mixer transmitting to a multiparty-unaware endpoint SHALL send
primary data only from one source per packet. The SSRC SHALL be the
SSRC of the mixer. The CSRC list MAY contain one member and be the
SSRC of the source of the primary data.
4.2.6. Functional Limitations
When a multiparty-unaware endpoint presents a conversation in one
display area in a chat style, it inserts source indications for
remote text and local user text as they are merged in completed text
groups. When an endpoint using this layout receives and presents
text mixed for multiparty-unaware endpoints, there will be two levels
of source indicators for the received text: one generated by the
mixer and inserted in a label after each source switch, and another
generated by the receiving endpoint and inserted after each switch
between the local source and the remote source in the presentation
area. This will waste display space and look inconsistent to the
reader.
New text can be presented from only one source at a time. Switching
the source to be presented takes place at suitable places in the
text, such as the end of a phrase, the end of a sentence, or a Line
Separator, or upon detecting inactivity. Therefore, the time to
switch to present waiting text from other sources may grow long, and
it will vary and depend on the actions of the currently presented
source.
Erasure can only be done up to the latest source switch. If a user
tries to erase more text, the erasing actions will be presented as a
letter "X" after the label.
Text loss because of network errors may hit the label between entries
from different parties, causing the risk of a misunderstanding
regarding which source provided a piece of text.
Because of these facts, it is strongly RECOMMENDED that multiparty
awareness be implemented in real-time text endpoints. The use of the
mixing method for multiparty-unaware endpoints should be left for use
with endpoints that are impossible to upgrade to become multiparty
aware.
4.2.7. Example Views of Presentation on Multiparty-Unaware Endpoints
The following pictures are examples of the view on a participant's
display for the multiparty-unaware case.
Figure 3 shows how a coordinated column view MAY be presented on
Alice's device in a view with two columns. The mixer inserts labels
to show how the sources alternate in the column with received text.
The mixer alternates between the sources at suitable points in the
text exchange so that text entries from each party can be
conveniently read.
___________________________________________________
| 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 3: Alice, Who Has a Conference-Unaware Client, Is
Receiving the Multiparty Real-Time Text in a Single Stream
In Figure 4, there is a tradition in receiving applications to
include a label showing the source of the text, here shown with
parentheses "()". The mixer also inserts source labels for the
multiparty call participants, here shown with brackets "[]".
_________________________________________________
| |^|
|(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 4: An Example of a View of the Multiparty-Unaware
Presentation in Chat Style, Where Alice Is the Local User
5. Relationship to Conference Control
5.1. Use with SIP Centralized Conferencing Framework
The Session Initiation Protocol (SIP) conferencing framework, mainly
specified in [RFC4353], [RFC4579], and [RFC4575], is suitable for
coordinating sessions, including multiparty real-time text. The
real-time text stream between the mixer and a participant is one and
the same during the conference. Participants get announced by
notifications when participants are joining or leaving, and further
user information may be provided. The SSRC of the text to expect
from joined users MAY be included in a notification. The
notifications MAY be used for both security purposes and translation
to a label for presentation to other users.
5.2. Conference Control
In managed conferences, control of the real-time text media SHOULD be
provided in the same way as for other media, e.g., for muting and
unmuting by the direction attributes in SDP [RFC8866].
Note that floor control functions may be of value for real-time text
users as well as for users of other media in a conference.
6. Gateway Considerations
Multiparty real-time text sessions may involve gateways of different
kinds. Gateways involved in setting up sessions SHALL correctly
reflect the multiparty capability or unawareness of the combination
of the gateway and the remote endpoint beyond the gateway.
6.1. Gateway Considerations with Textphones
One case that may occur is a gateway to the Public Switched Telephone
Network (PSTN) for communication with textphones (e.g., TTYs).
Textphones are limited devices with no multiparty awareness, and it
SHOULD therefore be appropriate for the gateway to not indicate
multiparty awareness for that case. Another solution is that the
gateway indicates multiparty capability towards the mixer and
includes the multiparty mixer function for multiparty-unaware
endpoints itself. This solution makes it possible to adapt to the
functional limitations of the textphone.
More information on gateways to textphones is found in [RFC5194].
6.2. Gateway Considerations with WebRTC
Gateway operation between RTP-mixer-based multiparty real-time text
and WebRTC-based real-time text may also be required. Real-time text
transport in WebRTC is specified in [RFC8865].
A multiparty bridge may have functionality for communicating via
real-time text in both (1) RTP streams with real-time text and (2)
WebRTC T.140 data channels. Other configurations may consist of a
multiparty bridge with either technology for real-time text transport
and a separate gateway for conversion of the text communication
streams between RTP and T.140 data channels.
In WebRTC, it is assumed that for a multiparty session, one T.140
data channel is established for each source from a gateway or bridge
to each participant. Each participant also has a data channel with a
two-way connection with the gateway or bridge.
A T.140 data channel used for two-way communication is for text from
the WebRTC user and from the bridge or gateway itself to the WebRTC
user. The label parameter of this T.140 data channel is used as the
NAME field in RTCP to participants on the RTP side. The other T.140
data channels are only for text from other participants to the WebRTC
user.
When a new participant has entered the session with RTP transport of
real-time text, a new T.140 data channel SHOULD be established to
WebRTC users with the label parameter composed of information from
the NAME field in RTCP on the RTP side.
When a new participant has entered the multiparty session with real-
time text transport in a WebRTC T.140 data channel, the new
participant SHOULD be announced by a notification to RTP users. The
label parameter from the WebRTC side or other suitable information
from the session or stream establishment procedure SHOULD be used to
compose the NAME RTCP field on the RTP side.
When a participant on the RTP side is disconnected from the
multiparty session, the corresponding T.140 data channel(s) SHOULD be
closed.
When a WebRTC user of T.140 data channels disconnects from the mixer,
the corresponding RTP streams or sources in an RTP-mixed stream
SHOULD be closed.
T.140 data channels MAY be opened and closed by negotiation or
renegotiation of the session, or by any other valid means, as
specified in Section 1 of [RFC8865].
7. Updates to RFC 4103
This document updates [RFC4103] by introducing an SDP media
attribute, "rtt-mixer", for negotiation of multiparty-mixing
capability with the format described in [RFC4103] and by specifying
the rules for packets when multiparty capability is negotiated and in
use.
8. Congestion Considerations
The congestion considerations and recommended actions provided in
[RFC4103] are also valid in multiparty situations.
The time values SHALL then be applied per source of text sent to a
receiver.
In the very unlikely event that many participants in a conference
send text simultaneously for a long period of time, a delay may build
up for the presentation of text at the receivers if the limitation in
characters per second ("cps") to be transmitted to the participants
is exceeded. A delay of more than 15 seconds can cause confusion in
the session. It is therefore RECOMMENDED that an RTP mixer discard
such text causing excessive delays and insert a general indication of
possible text loss [T140ad1] in the session. If the main text
contributor is indicated in any way, the mixer MAY avoid deleting
text from that participant. It should, however, be noted that human
creation of text normally contains pauses, when the transmission can
catch up, so that transmission-overload situations are expected to be
very rare.
9. IANA Considerations
9.1. Registration of the "rtt-mixer" SDP Media Attribute
IANA has registered the new SDP attribute "rtt-mixer".
Contact name: IESG
Contact email: iesg@ietf.org
Attribute name: rtt-mixer
Attribute semantics: See RFC 9071, Section 2.3
Attribute value: none
Usage level: media
Purpose: To indicate mixer and endpoint support of multiparty mixing
for real-time text transmission, using a common RTP stream for
transmission of text from a number of sources mixed with one
source at a time and where the source is indicated in a single
CSRC-list member.
Charset Dependent: no
O/A procedures: See RFC 9071, Section 2.3
Mux Category: NORMAL
Reference: RFC 9071
10. Security Considerations
The RTP-mixer model requires the mixer to be allowed to decrypt,
pack, and encrypt secured text from conference participants.
Therefore, the mixer needs to be trusted to maintain confidentiality
and integrity of the real-time text data. This situation is similar
to the situation for handling audio and video media in centralized
mixers.
The requirement to transfer information about the user in RTCP
reports in SDES, CNAME, and NAME fields, and in conference
notifications, may have privacy concerns, as already stated in RFC
3550 [RFC3550], and may be restricted for privacy reasons. When used
for the creation of readable labels in the presentation, the
receiving user will then get a more symbolic label for the source.
The services available through the real-time text mixer may be of
special interest to deaf and hard-of-hearing individuals. Some users
may want to refrain from revealing such characteristics broadly in
conferences. Conference systems where the mixer is included MAY need
to be designed with the confidentiality of such characteristics in
mind.
Participants with malicious intentions may appear and, for example,
disrupt the multiparty session by emitting a continuous flow of text.
They may also send text that appears to originate from other
participants. Countermeasures should include requiring secure
signaling, media, and authentication, and providing higher-layer
conference functions, e.g., for blocking, muting, and expelling
participants.
Participants with malicious intentions may also try to disrupt the
presentation by sending incomplete or malformed control codes.
Handling of text from the different sources by the receivers MUST
therefore be well separated so that the effects of such actions only
affect text from the source causing the action.
Care should be taken to avoid the possibility of attacks by
unauthenticated call participants, and even eavesdropping and
manipulation of content by non-participants, if the use of the mixer
is permitted for users both with and without security procedures.
As already stated in Section 3.18, security in media SHOULD be
applied by using DTLS-SRTP [RFC5764] at the media level.
Further security considerations specific to this application are
specified in Section 3.18.
11. References
11.1. Normative References
[ECMA-48] Ecma International, "ECMA-48: Control functions for coded
character sets", 5th edition, June 1991,
<https://www.ecma-international.org/publications-and-
standards/standards/ecma-48/>.
[ISO6429] ISO/IEC, "Information technology - Control functions for
coded character sets", ISO/IEC ISO/IEC 6429:1992, December
1992, <https://www.iso.org/obp/ui/#iso:std:iso-
iec:6429:ed-3:v1:en>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC4102] Jones, P., "Registration of the text/red MIME Sub-Type",
RFC 4102, DOI 10.17487/RFC4102, June 2005,
<https://www.rfc-editor.org/info/rfc4102>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630,
DOI 10.17487/RFC5630, October 2009,
<https://www.rfc-editor.org/info/rfc5630>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011,
<https://www.rfc-editor.org/info/rfc6263>.
[RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
Thomson, "Session Traversal Utilities for NAT (STUN) Usage
for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
October 2015, <https://www.rfc-editor.org/info/rfc7675>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text
Conversation over WebRTC Data Channels", RFC 8865,
DOI 10.17487/RFC8865, January 2021,
<https://www.rfc-editor.org/info/rfc8865>.
[RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP:
Session Description Protocol", RFC 8866,
DOI 10.17487/RFC8866, January 2021,
<https://www.rfc-editor.org/info/rfc8866>.
[T140] ITU-T, "Protocol for multimedia application text
conversation", ITU-T Recommendation T.140, February 1998,
<https://www.itu.int/rec/T-REC-T.140-199802-I/en>.
[T140ad1] ITU-T, "Recommendation T.140 Addendum", February 2000,
<https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>.
11.2. Informative References
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353,
DOI 10.17487/RFC4353, February 2006,
<https://www.rfc-editor.org/info/rfc4353>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <https://www.rfc-editor.org/info/rfc4575>.
[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol
(SIP) Call Control - Conferencing for User Agents",
BCP 119, RFC 4579, DOI 10.17487/RFC4579, August 2006,
<https://www.rfc-editor.org/info/rfc4579>.
[RFC5194] van Wijk, A., Ed. and G. Gybels, Ed., "Framework for Real-
Time Text over IP Using the Session Initiation Protocol
(SIP)", RFC 5194, DOI 10.17487/RFC5194, June 2008,
<https://www.rfc-editor.org/info/rfc5194>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC8643] Johnston, A., Aboba, B., Hutton, A., Jesske, R., and T.
Stach, "An Opportunistic Approach for Secure Real-time
Transport Protocol (OSRTP)", RFC 8643,
DOI 10.17487/RFC8643, August 2019,
<https://www.rfc-editor.org/info/rfc8643>.
[RFC8723] Jennings, C., Jones, P., Barnes, R., and A.B. Roach,
"Double Encryption Procedures for the Secure Real-Time
Transport Protocol (SRTP)", RFC 8723,
DOI 10.17487/RFC8723, April 2020,
<https://www.rfc-editor.org/info/rfc8723>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
Acknowledgements
The author wants to thank the following persons for support, reviews,
and valuable comments: Bernard Aboba, Amanda Baber, Roman Danyliw,
Spencer Dawkins, Martin Duke, Lars Eggert, James Hamlin, Benjamin
Kaduk, Murray Kucherawy, Paul Kyzivat, Jonathan Lennox, Lorenzo
Miniero, Dan Mongrain, Francesca Palombini, Colin Perkins, Brian
Rosen, Rich Salz, Jürgen Schönwälder, Robert Wilton, Dale Worley,
Yong Xin, and Peter Yee.
Author's Address
Gunnar Hellström
Gunnar Hellström Accessible Communication
SE-13670 Vendelsö
Sweden
Email: gunnar.hellstrom@ghaccess.se