| Internet Engineering Task Force | G. Hellstrom |
| Internet-Draft | Omnitor |
| Intended status: Best Current Practice | March 3, 2020 |
| Expires: September 4, 2020 |
Real-time text media handling in multi-party conferences
draft-hellstrom-mmusic-multi-party-rtt-02
This memo specifies methods for Real-Time Text (RTT) media handling in multi-party calls. The main RTT transport is to carry Real-Time text by the RTP protocol in a time-sampled mode according to RFC 4103 RFC 4103 . The mechanisms enable the receiving application to present the received real-time text medium separated per source, in different ways according to user preferences. Some presentation related features are also described explaining suitable variations of transmission and presentation of text.
Call control features are described for the SIP environment. A number of alternative methods for providing the multi-party negotiation, transmission and presentation are discussed and a recommendation for the main one is provided. 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 stream.
Alternative methods using a single RTP stream and source identification inline in the text stream are also described, one of them being provided as a lower functionality fallback method for endpoints with no multi-party awareness for RTT.
Bridging methods where the text stream is carried untouched by the bridge are also discussed.
Brief information is also provided for multi-party RTT in the WebRTC environment.
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Real-time text (RTT) is a medium in real-time conversational sessions. Text entered by participants in a session is transmitted in a time-sampled fashion, so that no specific user action is needed to cause transmission. This gives a direct flow of text in the rate it is created, that is suitable in a real-time conversational setting. The real-time text medium can be combined with other media in multimedia sessions.
Media from a number of multimedia session participants can be combined in a multi-party session. This memo specifies how the real-time text streams can be handled in multi-party sessions.
The description is mainly focused on the transport level, but also describes a few session and presentation level aspects.
Transport of real-time text is specified in RFC 4103 RTP Payload for text conversation. It makes use of RFC 3550 Real Time Protocol, for transport. Robustness against network transmission problems is normally achieved through redundant 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 RFC 3261 environment.
A very brief overview of functions for real-time text handling in multi-party sessions is described in RFC 4597 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.
Real-time text can also be transported in the WebRTC environment, by using WebRTC data channels according to [I-D.ietf-mmusic-t140-usage-data-channel]. Multi-party aspects for WebRTC solutions are briefly covered.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119.
In the centralized conference model for SIP, introduced in RFC 4353 A Framework for Conferencing with the Session Initiation Protocol (SIP), one function co-ordinates the communication with participants in the multi-party session. This function also controls media mixer functions for the media appearing in the session. The central function is common for control of all media, while the media mixers may work differently for each media.
The central function is called the Focus UA. Many variants exist for setting up sessions including the multipoint control centre. It is not within scope of this description to describe these, but rather the media specific handling in the mixer required to handle multi-party calls with RTT.
The main principle for handling real-time text media in a centralized conference is that one RTP session for real-time text is established including the multipoint media control centre and the participating endpoints which are going to have real-time text exchange with the others.
The different possible mechanisms for mixing and transporting RTT differs in the way they multiplex the text streams and how they identify the sources of the streams. RFC 7667 describes a number of possible use cases for RTP. This specification refers to different sections of RFC 7667 for further reading of the situations caused by the different possible design choices.
The recommended method for using RTT in a centralized conference model is specified in [I-D.hellstrom-avtcore-multi-party-rtt-source]
Real-time text can also be transported in the WebRTC environment, by using WebRTC datachannels according to [I-D.ietf-mmusic-t140-usage-data-channel]. Ways to handle multi-party calls in that environmnent are also specified.
The following requirements are placed on multi-party RTT:
Coordinating and sending text RTP streams in the multi-party session can be done in a number of ways. The most suitable methods are specified here with pros and cons.
A receiving and presenting UA SHOULD separate text from the different sources and identify and display them accordingly.
A set of solutions can be based on the central RTP mixer. They are described here and a preferred method selected.
An RTP media mixer combines text from participants into one RTP stream , thus all using the same destination address/port combination, the same RTP SSRC and , one sequence number series as described in Section 7.1 and 7.3 of RTP RFC 3550 about the Mixer function. This method is also briefly described in RFC 7667, section 3.6.1 Media mixing mixer [RFC7667].
The sources of the text in each RTP packet are identified by the CSRC list in the RTP packets, containing the SSRC of the initial sources of text. The order of the CSRC parameters is with the SSRC of the source of the primary text first, followed by the SSRC of the first level redundancy, and then the second level.
A set of specific rules for the application of this method together with RFC 4103 [RFC4103]is specified in [I-D.hellstrom-avtcore-multi-party-rtt-source]
The identification of the sources is made through the CSRC fields and can be made more readable through the RTCP SDES CNAME and NAME packets as described in RTP[RFC3550].
Also information provided through the notification according to RFC 4575 [RFC4575] when the participant joined the conference provides suitable information and a reference to the SSRC.
A receiving UA is supposed to separate text items from the different sources and identify and display them accordingly.
The ordered CSRC lists in the RFC 4103 [RFC4103] packets make it possible to recover from loss of one and two packets in sequence and assign the recovered text to the right source. For more loss, a marker for possible loss should be inserted or presented.
The conference server 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.
Pros:
This method has low overhead.
When loss of packets occur, it is possible to recover text from redundancy at loss of up to the number of redundancy levels carried in the RFC 4103 [RFC4103] stream. (normally primary and two redundant levels.
This method can be implemented with most RTP implementations.
Cons:
When more consecutive packet loss than the number of generations of redundant data appears, it is not possible to deduct the sources of the totally lost data.
The conference server need to be allowed to decrypt/encrypt the packet payload. This is however normal for media mixers for other media.
Text from all participants except the receiving one is transmitted from the media mixer in the same RTP session and stream, thus all using the same destination address/port combination, the same RTP SSRC and , one sequence number series as described in Section 7.1 and 7.3 of RTP RFC 3550 about the Mixer function. The sources of the text in each RTP packet are identified by a new defined T.140 control code "c" followed by a unique identification of the source in UTF-8 string format.
The receiver can use the string for presenting the source of text. This method is on the RTP level described in RFC 7667, section 3.6.1 Media mixing mixer [RFC7667].
The inline coding of the source of text is applied in the data stream itself, and an RTP mixer function is used for coordinating the sources of text into one RTP stream.
Information uniquely identifying each user in the multi-party session is placed as the parameter value “n” in the T.140 application protocol function with the function code “c”. The identifier shall thus be formatted like this: SOS c n ST, where SOS and ST are coded as specified in ITU-T T.140. The "c" is the letter "c". The n parameter value is a string uniquely identifying the source. This parameter shall be kept short so that it can be repeated in the transmission without concerns for network load.
A receiving UA is supposed to separate text items from the different sources and identify and display them accordingly.
The conference server need to be allowed to decrypt/encrypt the packet payload in order to check the source and repack the text.
Pros:
If loss of packets occur, it is possible to recover text from redundancy at loss of up to the number of redundancy levels carried in the RFC 4103 [RFC4103]stream. (normally primary and two redundant levels.
This method can be implemented with most RTP implementations.
Transmitted text can also be used with other transports than RTP
Cons:
The method implies a moderate load by the need to insert the source often in the stream.
If more consecutive packet loss than the number of generations of redundant data appears, it is not possible to deduct the source of the totally lost data.
The mixer needs to be able to generate suitable and unique source identifications which are suitable as labels for the sources.
Requires an extension on the ITU-T T.140 standard, best made by the ITU.
The conference server need to be allowed to decrypt/encrypt the packet payload.
The conference server need to be allowed to decrypt/encrypt the packet payload.
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.
This method MAY be used when no support for multi-party awareness is detected in the receiving endpoint.The base for his method is described in RFC 7667, section 3.6.1 Media mixing mixer [RFC7667].
See Appendix A for an informative example of a procedure for presenting RTT to a conference-unaware UA.
Pros:
Can be transmitted to conference-unaware endpoints.
Can be used with other transports than RTP
Cons:
Does not allow full real-time presentation of more than one source at a time. Text from other sources will be delayed, even if automatic detection of suitable moments for switching source for presentation is made by the mixer.
The only realistic presentation format is a style with the text from the different sources presented with a text label indicating source, and the text collected in a chat style presentation but with more frequent turn-taking.
Endpoints often have their own system for adding labels to the RTT presentation. In that case there will be two levels of labels in the presentation, one for the mixer and one for the sources.
If loss of more packets than can be recovered by the redundancy appears, it is not possible to detect which source was struck by the loss. It is also possible that a source switch occurred during the loss, and therefore a false indication of the source of text can be provided to the user after such loss.
Because of all these cons, this method is not recommended and MUST NOT be used as the main method, but only as the last resort for backwards interoperability with conference-unaware endpoints.
The conference server need to be allowed to decrypt/encrypt the packet payload.
It may be desirable to send text in a multi-party setting in a way that allows the text stream contents to be distributed without decryption and encryption in any central server. A number of such methods are described. However, when writing this specification, no one of these methods have a specified way of establishing the session by sdp.
Within the RTP session, text from each participant is transmitted from the RTP media translator in a separate RTP stream, thus using the same destination address/port combination, but separate RTP SSRC parameters and sequence number series as described in Section 7.1 and 7.2 of RTP RFC 3550 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.
A receiving and presenting UA is supposed to separate text items from the different sources and identify and display them in a suitable way.
This method is described in RFC 7667, section 3.5.1 Relay-transport translator or 3.5.2 Media translator [RFC7667].
The identification of the source is made through the RTCP SDES CNAME and NAME packets as described in RTP[RFC3550].
Pros:
This method has moderate overhead. When loss of packets occur, it is possible to recover text from redundancy at loss of up to the number of redundancy levels carried in the RFC 4103 [RFC4103] stream. (normally primary and two redundant levels.
More loss than what can be recovered, can be detected and the marker for text loss can be inserted in the correct stream.
It may be possible in some scenarios to keep the text encrypted through the Translator.
Cons:
There may be RTP implementations not supporting the Translator model.
This configuration is not supported by current media declarations in sdp. RFC 3264 [RFC3264]specifies in many places that one media description is supposed to describe just one RTP stream.
In order to achieve end-to-end encryption, it is possible to let the packets from the sources just pass though a central distributor, and handle the security agreements between the participants. Specifications exist for a framework with this functionality suitable for application on RTP based conferences in [I-D.ietf-perc-private-media-framework]. The RTP flow and mixing characteristics has similarities with the method described under "RTP Translator sending one RTT stream per participant" above. RFC 4103 RTP streams [RFC4103] would fit into the structure and it would provide a base for end-to-end encrypted rtt multi-party conferencing.
Pros:
Good security
Straightforward multi-party handling.
Cons:
Does not operate under the usual SIP central conferencing architecture.
Requires the participants to perform a lot of key handling.
Text from all participants are transmitted directly to all others in one RTP session, without a central bridge. The sources of the text in each RTP packet are identified by the source network address and the SSRC.
This method is described in RFC 7667, section 3.4 Point to multi-point using mesh [RFC7667].
Pros:
When loss of packets occur, it is possible to recover text from redundancy at loss of up to the number of redundancy levels carried in the RFC 4103 [RFC4103] stream. (normally primary and two redundant levels.
This method can be implemented with most RTP implementations.
Transmitted text can also be used with other transports than RTP
Cons:
This model is not described in IMS, NENA and EENA specifications, and does therefore not meet the requirements.
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.
This method is out of scope for further discussion here, because the foreseen applications use centralized model conferencing.
Pros:
When loss of packets occur, it is possible to recover text from redundancy at loss of up to the number of redundancy levels carried in the RFC 4103 [RFC4103] stream. (normally primary and two redundant levels.
Complete loss of text can be indicated in the received stream.
This method can be implemented with most RTP implementations.
End-to-end encryption is achievable.
Cons:
This method is not described in IMS, NENA and ETSI specifications and does therefore not meet the requirements.
A lot of network resources are spent on setting up separate sessions for each participant.
Within WebRTC, real-time text is specified to be carried in WebRTC data channels as specified in [I-D.ietf-mmusic-t140-usage-data-channel]. A few ways to handle multi-party RTT are mentioned briefly. They are explained and further detailed below.
A straightforward way to handle multi-party RTT is for the bridge to open one T.140 data channel per source towards the receiving participants.
The stream-id forms a unique stream identification.
The identification of the source is made through the Label property of the channel, and session information belonging to the source. The UA can compose a readable label for the presentation from this information.
Pros:
This is a straightforward solution.
Cons:
With a high number of participants, the overhead of establishing the high number of data channels required may be high.
A way to handle multi-party RTT in WebRTC is for the bridge combine text from all sources into one data channel and insert the sources in the stream by a T.140 control code for source.
This method is described in a corresponding section for RTP transmission above Section 4.1.2.
The identification of the source is made through insertion in the beginning of each text transmission from a source of a control code extension "c" followed by a string representing the source, framed by the control code start and end flags SOS and ST (See ITU-T T.140).
A receiving UA is supposed to separate text items from the different sources and identify and display them in a suitable way.
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.
Pros:
This solution has relatively low overhead on session and network level
Cons:
This solution has higher overhead on the media contents level than the WebRTC solution above.
Standardisation of the new control code "c" in ITU-T T.140 [T140] is required.
The conference server need to be allowed to decrypt/encrypt the data channel contents.
For RTP transport of RTT using RTP-mixer technology, one method for multi-party mixing and transport stand out as fulfilling the goals best and is therefore recommended. That is: "RTP Mixer indicating participants in CSRC list" Section 4.1.1.
For RTP transport in separate streams or sessions, a bridging method with good characteristics is the end-to-end encryption model "perc" Section 4.2.2.
For WebRTC, one method is to prefer because of the simplicity. So, for WebRTC, the method to implement for multi-party RTT with conference-aware parties when no other method is explicitly agreed between implementing parties is: "RTT bridging in WebRTC with one data channel per source" Section 4.3.1.
General session control aspects for multi-party sessions are described in RFC 4575 A Session Initiation Protocol (SIP) Event Package for Conference State, and RFC 4579 Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents. The nomenclature of these specifications are used here.
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 negotiated.
A method for detection of conference-awareness for centralized SIP conferencing in general is specified in RFC 4579. 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 [RFC4575]. The mechanism is described in detail in RFC 4575.
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 of the centralized conference model according to this specification.
Capability for RTT multi-party handling can be decided to be implicitly indicated by session control items.
The focus may implicitly indicate muti-party RTT capability by including the media child with value "text" in the RFC 4575 [RFC4575] conference-info provided in conference notifications.
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.
The implicit RTT capability indication means for the focus that it can handle multi-party RTT according to the preferred method indicated in the RTT multi-party methods section above.
The implicit RTT capability indication means for the UA that it can handle multi-party RTT according to the preferred method indicated in the RTT multi-party methods section above.
If the focus detects that a UA implicitly declared RTT multi-party capability, it SHALL provide RTT according to the preferred method.
If the focus detects that the UA does not indicate any RTT multi-party capability, then it shall either provide RTT multi-party text in the way specified for conference-unaware UA above, or refuse to set up the session.
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.
Pros:
Acceptance of implicit multi-party capability implies that no standardisation of explicit RTT multi-party capability exchange is required.
Cons:
If other methods for multi-party RTT are to be used in the same implementation environment as the preferred ones,then capability exchange needs to be defined for them.
Cannot be used outside a strictly applied SIP central conference model.
Specifications for RTT multi-party capability declarations can be agreed for use as SIP media feature tags, to be exchanged during SIP call control operation according to the mechanisms in RFC 3840 [RFC3840] and RFC 3841 [RFC3841]. Capability for the RTT Multi-party capability is then indicated by the media feature tag "rtt-mix", with a set of possible values for the different possible methods.
The possible values in the list may be:
rtp-mixer indicates capability for using the RTP-mixer based presentation of multi-party text.
perc indicates capability for using the perc based transmission of multi-party text.
Example: Contact: <sip:a2@beco.example.com>
;methods="INVITE,ACK,OPTIONS,BYE,CANCEL"
;+sip.rtt-mix="rtp-mixer"
If, after evaluation of the alternatives in this specification, only one mixing method is selected to be brought to implementation, then the media tag can be reduced to a single tag with no list of values.
An offer-answer exchange should take place and the common method selected by the answering party shall be used in the session with that UA.
When no common method is declared, then only the fallback method for multi-party unaware participants can be used, or the session dropped.
If more than one text media line is included in SDP, all must be capable of using the declared RTT multi-party method.
Pros:
Provides a clear decision method.
Can be extended with new mixing methods.
Can guide call routing to a suitable capable focus.
Cons:
Requires standardization and IANA registration.
Is not stream specific. If more than one text stream is specified, all must have the same type of multi-party capability.
Cannot be used in the WebRTC environment.
An attribute can be specified on media level, to be used in text media SDP declarations for negotiating RTT multi-party capabilities. The attribute can have the name "rtt-mix", with one or more of its possible values in a comma-separated list.
The possible values in the list are:
rtp-mixer indicates capability for using the RTP-mixer based presentation of multi-party text.
perc indicates capability for using the perc based transmission of multi-party text.
An offer-answer exchange should take place and the common method selected by the answering party shall be used in the session with that UA.
When no common method is declared, then only the fallback method can be used.
Example: a=rtt-mix:rtp-mixer
If, after evaluation of the alternatives in this specification, only one mixing method is selected to be brought to implementation, then the attribute can be reduced to a single attribute with no list of values.
Pros:
Provides a clear decision method.
Can be extended with new mixing methods.
Can be used on specific text media.
Can be used also for SDP-controlled WebRTC sessions with multiple streams in the same data channel.
Cons:
Requires standardization and IANA registration.
Cannot guide SIP routing.
An attribute can be specified on media level, to be used in text media SDP declarations for negotiating RTT multi-party capabilities. The attribute can have the name "rtt-mix" with no value. It would be selected and used if only one method for multi-party rtt is brought forward from this specification, and the other suppressed or found to be possible to negotiate in another way..
An offer-answer exchange should take place and if both parties specify rtt-mix capability, the method for indicating source in the CSRC-list shall be used.
When no common method is declared, then only the fallback method can be used, or the session not accepted for multi-party use.
Example: a=rtt-mix
Pros:
Provides a clear decision method.
Very simple syntax and semantics.
Can be used on specific text media.
Can be used also for SDP-controlled WebRTC sessions with multiple streams in the same data channel.
Cons:
Requires standardization and IANA registration.
Cannot guide SIP routing.
An FMTP format parameter can be specified for the RFC 4103 [RFC4103]media, to be used in text media SDP declarations for negotiating RTT multi-party capabilities. The parameter can have the name "rtt-mix", with one or more of its possible values.
The possible values in the list are:
rtp-mixer indicates capability for using the RTP-mixer based presentation of multi-party text.
perc indicates capability for using the perc based transmission of multi-party text.
Example: a=fmtp 96 98/98/98 cps=30;rtt-mix=rtp-mixer
If, after evaluation of the alternatives in this specification, only one mixing method is selected to be brought to implementation, then the parameter can be reduced to a single parameter with no list of values.
An offer-answer exchange should take place and the common method selected by the answering party shall be used in the session with that UA.
When no common method is declared, then only the fallback method can be used, or the session denied.
Pros:
Provides a clear decision method.
Can be extended with new mixing methods.
Can be used on specific text media.
Can be used also for SDP-controlled WebRTC sessions with multiple streams in the same data channel.
Cons:
Requires standardization and IANA registration.
May cause interop problems with current RFC4103 [RFC4103] implementations not expecting a new fmtp-parameter.
Cannot guide SIP routing.
The preferred capability declaration method is the one with a simplified SDP attribute "a=rtt-mix" Section 6.4 because it is straightforward and partially usable also for WebRTC.
The main way to identify the source of text in the RTP based solution is by the SSRC of the sending participant. In the RTP-mixer solution, it is included in the CSRC list of the transmitted packets. Further identification that may be needed for better labeling of received text may be achieved from a number of sources. It may be the RTCP SDES CNAME and NAME reports, and in the conference notification data (RFC 4575) [RFC4575].
As soon as a new member is added to the RTP session, its characteristics should be transmitted in RTCP SDES CNAME and NAME reports according to section 6.5 in RFC 3550 [RFC3550]. The information about the participant should also be included in the conference data including the text media member in a notification according to RFC 4575 [RFC4575].
The RTCP SDES report, SHOULD contain identification of the source represented by the SSRC/CSRC identifier. This identification MUST contain the CNAME field and MAY contain the NAME field and other defined fields of the SDES report.
A focus UA SHOULD primarily convey SDES information received from the sources of the session members. When such information is not available, the focus UA SHOULD compose SSRC/CSRC, CNAME and NAME information from available information from the SIP session with the participant.
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.
A source identity SHOULD be composed from available information sources and displayed together with the text as indicated in ITU-T T.140 Appendix[T140].
The source identity should primarily be the NAME field from incoming SDES packets. If this information is not available, and the session is a two-party session, then the T.140 source identity SHOULD be composed from the SIP session participant information. For multi-party sessions the source identity may be composed by local information if sufficient information is not available in the session.
Applications may abbreviate the presented source identity to a suitable form for the available display.
Applications may also replace received source information with internally used nicknames.
The multi-party aware UA should after any action for recovery of data from lost packets, separate the incoming streams and present them according to the style that the receiving application supports and the user has selected. The decisions taken for presentation of the multi-party interchange shall be purely on the receiving side. The sending application must not insert any item in the stream to influence presentation that is not requested by the sending participant.
One often used style is to present real-time text in chunks in readable bubbles identified by labels containing names of sources. Bubbles are placed in one column in the presentation area and are closed and moved upwards in the presentation area after certain items or events, when there is also newer text from another source that would go into a new bubble. The text items that allows bubble closing are any character closing a phrase or sentence followed by a space or a timeout of a suitable time (about 10 seconds).
Real-time active text sent from the local user should be presented in a separate area. When there is a reason to close a bubble from the local user, the bubble should be placed above all real-time active bubbles, so that the time order that real-time text entries were completed is visible.
Scrolling is usually provided for viewing of recent or older text. When scrolling is done to an earlier point in the text, the presentation shall not move the scroll position by new received text. It must be the decision of the local user to return to automatic viewing of latest text actions. It may be useful with an indication that there is new text to read after scrolling to an earlier position has been activated.
The presentation area may become too small to present all text in all real-time active bubbles. Various techniques can be applied to provide a good overview and good reading opportunity even in such situations. The active real-time bubble may have a limited number of lines and if their contents need more lines, then a scrolling opportunity within the real-time active bubble is provided. Another method can be to only show the label and the last line of the active real-time bubble contents, and make it possible to expand or compress the bubble presentation between full view and one line view.
Erasures require special consideration. Erasure within a real-time active bubble is straightforward. But if erasure from one participant affects the last character before a bubble, the whole previous bubble becomes the actual bubble for real-time action by that participant and is placed below all other bubbles in the presentation area. If the border between bubbles was caused by the CRLF characters (instead of the normal "Line Separator"), only one erasure action is required to erase this bubble border. When a bubble is closed, it is moved up, above all real-time active bubbles.
A three-party view is shown in this example .
_________________________________________________
| |^|
| | |
| | |
| | |
|[Alice] Hi, Alice here. | |
| | |
|[Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris. | |
| I thought you should be here. | |
| | |
|[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|[Bob] And I on Wednesday evening. | |
| | |
|[Alice] Can we meet on Thursday evening? | |
| | |
|[Eve] Yes, definitely. How about 7pm. | |
| at the entrance of the restaurant | |
| Le Lion Blanc? | |
|[Eve] we can have dinner and then take a walk | |
| | |
| <Eve-typing> But I need to be back to | |
| the hotel by 11 because I need | |
| |-|
| <Bob-typing> I wou |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 1: Example of a three-party call presented in the bubble style.
Figure 1: Three-party call with bubble style.
Other presentation styles than the bubble style may be arranged and appreciated by the users. In a video conference one way may be to have a real-time text area below the video view of each participant. Another view may be to provide one column in a presentation area for each participant and place the text entries in a relative vertical position corresponding to when text entry in them was completed. The labels can then be placed in the column header. The considerations for ending and moving and erasure of entered text discussed above for the bubble style are valid also for these styles.
This figure shows how a coordinated column view MAY be presented.
_____________________________________________________________________ | Bob | Eve | Alice | |____________________|______________________|_______________________| | | |I will arrive by TGV. | |My flight is to Orly| |Convenient to the main | | |Hi all, can we plan |station. | | |for the seminar? | | |Eve, will you do | | | |your presentation on| | | |Friday? |Yes, Friday at 10. | | |Fine, wo | |We need to meet befo | |___________________________________________________________________|
Figure 2: A coordinated column-view of a three-party session with entries ordered in approximate time-order.
Multi-party unaware UA:s are prepared only for presentation of two sources of text, the local user and a remote user. If mixing for multi-party unaware UAs is to be supported, in order to enable some multi-party communication with such UA, the mixer need to plan the presentation and insert labels and line breaks before lables. Many limitations appear for this presentation mode, and it must be seen as a fallback and a last resort. A realistic alternative is to not allow multi-party sessions with multi-party unaware UAs.
See Appendix A for an informative example of a procedure for presenting RTT to a conference-unaware UA.
The security considerations valid for RFC 4103 [RFC4103] and RFC 3550 [RFC3550] are valid also for the multi-party sessions with text.
The items for indication and negotiation of capability for multi-party rtt should be registered with IANA in the specifications where they are specified in detail.
The congestion considerations described in RFC 4103 [RFC4103] 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.
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.
Arnoud van Wijk for contributions to an earlier, expired draft of this memo.
Change from a general overview to overview with clear recommendations.
Splits text coordination methods in three groups.
Recommends rtt-mixer with sources in CSRC-list but referenes to its spec for details.
Shortened Appendix with conference-unaware example.
Cleaned up preferences.
Inserted pictures of screen-views.
| [RFC2119] | Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997. |
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.
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.
The procedures may also be applied on a terminal for display of multiple streams of real-time text in one area.
The intention is that text from each participant shall be displayed in suitable sections so that 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.
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.
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.
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.
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.
The conference server applies these mixing procedures to text transmitted to call participants who have not gone through a completed negotiation for conference awareness in real-time text presentation.
All the participants and the conference server use real-time text conversation presentation coding according to ITU-T T.140 [T.140]. A consequence is that real-time text transmissions are UTF-8 coded, with control codes selected from ISO 6429 [ISO 6429].
The description is from the conference server point of view.
The real-time text mixer structure described here is supposed to be placed in the media path so that it is implemented with one mixer per recipient. A mixer contains buffers for temporary storage of text intended for the recipient. Each mixer has one buffer for each contributing participant. A set of status variables is maintained per buffer and is used in the mixer actions. The mixer logic decides for each moment which participant?s buffer content is to be sent on to the recipient. By default, the recipient does not contribute text to its own mixer. Text transmitted by a participant is usually displayed locally and it will only cause confusion if it appears also in received text.
This description of the mixer is valid per recipient.
Text from each contributing participant is checked for a set of characteristics on reception.
The following pictures are examples of the view on a participant's display.
_________________________________________________ | Conference | Alice | |________________________|_________________________| | |I will arrive by TGV. | |[Bob]:My flight is to |Convenient to the main | |Orly. |station. | |[Eve]:Hi all, can we | | |plan for the seminar. | | | | | |[Bob]:Eve, will you do | | |your presentation on | | |Friday? | | |[Eve]:Yes, Friday at 10.| | |[Bob]: Fine, wo |We need to meet befo | |________________________|_________________________|
Figure A1 : Alice who has a conference-unaware client is receiving the multi-party real-time text in a single-stream. This figure shows how a coordinated column view MAY be presented on Alice's device.
_________________________________________________
| |^|
|[mix][Alice] Hi, Alice here. | |
| | |
|[mix][Bob] Bob as well. | |
| | |
|[mix][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. | |
| | |
|[mix][Eve] we can have dinner and then walk | |
| | |
|[mix][Eve] But I need to be back to | |
| the hotel by 11 because I need |-|
| |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure A2 shows a conference view with real-time text preview. Bob's text is buffering until a Current switch reason.
This appendix was developed with funding in part from the National Institute on Disability and Rehabilitation Research, U.S. Department of Education,RERC on Telecommunications Access,?grant # H133E090001?. However, the contents do not necessarily represent the policy of the Department of Education, and you should not assume endorsement by the Federal Government.