FW: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-media-content-04.txt
"Jean-Francois Mule" <jf.mule@cablelabs.com> Mon, 24 July 2006 06:56 UTC
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Subject: FW: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-media-content-04.txt
Date: Mon, 24 Jul 2006 00:56:23 -0600
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From: Jean-Francois Mule <jf.mule@cablelabs.com>
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Large msg from Gunnar converted to plain txt. -----Original Message----- From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] Sent: Monday, July 24, 2006 12:53 AM To: Jean-Francois Mule Subject: Forward of moderated message -----Original Message----- From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se] Sent: Sunday, July 23, 2006 3:27 PM To: Gunnar Hellstrom; Arnoud van Wijk; 'Colin Perkins' Cc: mmusic@ietf.org Subject: RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-media-content-04.txt Colin, I assume that it was the talk about half-duplex in the V.151 Annex that made you hesitate about the appropriateness of the content parameter "txp". That was taken from PSTN text-telephony jargon. I have adjusted the V.151 Annex proposal below to properly describe the use of the "txp" parameter in that application in plain user terms. So, I instead now say "The line "a=content:txp" is sent from the gateway as an indication to the ITD user that there is a likely need for the user to apply the formal turntaking habits that are common between PSTN text telephone users. " If you want, we can again refine the description of the "txp" parameter in the media-content draft, section 5. Current wording is: txp: This indicates that the media stream is originated from a textphone that is incapable of handling simultaneous two-way communication. This limitation requires special behavior from the user of a terminal receiving the indication. The indication should be used for an indication in the user interface. A typical use case is a connection where one endpoint is an analog textphone of a kind that cannot handle two-way simultaneous text communication, and the other one is a native IP based real time text capable terminal. The human user of the IP terminal need to change behaviour when this indication is received, and apply formal turn-taking habits. They may also need to figure out to what extent it is possible to interrupt the other party if the need arises. My suggestion, if you find it better: txp: This indicates that the media stream is originated from a textphone with some presentation limitation. This limitation makes it probable that the user should apply formal turntaking habits that are common among text telephone users in the PSTN. The indication should be used for an indication in the user interface. A typical use case is a connection where one endpoint is an analog textphone of a kind that merges text from both ends in the same window, and the other one is a native IP based real time text capable terminal. The human user of the IP terminal need to change behaviour when this indication is received, and apply formal turn-taking habits. They may also need to figure out to what extent it is possible to interrupt the other party if the need arises, because that possibility varies between textphone types. ----------------------Proposed alternative to ITU-T V.151 Annex D, using caller and callee capabilities and preferences and media-contents to control gateway - IP text capable terminal communication----------------------------------------------------------- ------------- Alternative proposal for V.151 Annex D. ANNEX D2 Interworking IP Text Devices with V.151 Gateways. Connect text channel when usage of real-time text is likely. D.1 Introduction Some non-Gateway IP devices in the network not having a means of modulating or demodulating PSTN textphone signals, such as IVR systems, voicemail systems, IP phones, Multimedia telephones or other devices, may support the transmission of real-time text over IP networks and, ideally, interwork with V.151-compliant Gateways in order to provide a means through which users of PSTN textphone devices can communicate with those IP devices. This class of devices is referred to herein as an "IP Text Device" (ITD). This Annex defines the procedures that may be used in order to allow for such interworking between V.151 Gateways and ITDs. The main principle is to use RFC 4103 as the default text transport method between gateways and ITD:s, and open text channels only when it is likely that the text channel will be used. D.2 Exchanging capabilities and opening media streams When establishing a call, an ITD may advertise support for real-time text in accordance with IETF RFC 4103. In general, ITDs also transmit text characters when communicating with other ITDs using RFC 4103, which specifies the establishment of a separate RTP stream specifically for transmitting text characters. However, since V.151 PSTN gateway devices interleave audio and text data when used for Gateway - to - gateway operation, V.151 gateways need to support both RFC 4103 to interwork with ITDs and RFC 4351 for communicating with other PSTN gateways. As such, V.151 gateways are Recommended to indicate support for both transport methods initially when exchanging capability information. However, relatively few calls use real-time text in the PSTN, and therefore it may be desirable for the gateway to not open the text channel until either party in the call explicitly acts to use real-time text. According to normal procedures in IP multimedia systems, it is allowable to indicate capability for other transport protocols for real-time text as well as RFC 4103 and connect such alternatives. D.2.1 SDP based systems Similar logic follows for SDP-based systems. In SIP-based systems, the terminal and gateway shall announce support for text in the Contact field according to RFC 3840, and may indicate preference for text according to RFC 3841. Contact: <uri>;text A calling SIP terminal who does not put priority to text may indicate an interest to use text by: Accept-Contact: *;text A calling SIP terminal who want to put priority to establishing a text connection shall indicate that by: Accept-Contact: *;text;require Most commonly, PSTN text telephones support alternating text and audio during the call, but not simultaneous text and audio. If the gateway only supports alternating text and audio during the PSTN text session, it may be beneficial to make the ITD user aware of this limitation. It is proposed to use the Duplex parameter of RFC 3840 for this purpose. Contact: duplex="half" There is a parameter also for use in the text media description: a=content:txp It should be used to indicate to the ITD user that there is a likely need to use the traditional formal turn-taking habits that are common among PSTN text telephone users.. Note that while the use of RFC 2198 redundancy or other fault tolerance scheme is not shown in the abbreviated examples below, appropriate mechanisms should be employed to protect the transmission of the text stream in accordance with the text transport specifications. D.2.1.1 Call from PSTN Consider the following abbreviated example SDP that might be found in an offer sent by a V.151 Gateway not offering RFC 4103 support initially, but indicating text support and that the text support is alternating with audio: Contact: <uri>;text;duplex="half" m=audio 7200 RTP/AVP 0 98 a=rtpmap:98 t140c/8000 a=fmtp:98 cps=20 If the gateway at that time already knows that text will be needed in the call, a requirement for text shall be included, and a media line for text. The decision to include text when offering the call may be because textphone tones have already been detected on the PSTN line, the gateway may be configured to be a dedicated text gateway, or it may be known by subscription or any other external means that either user has preference for text. Accept-Contact: *;text;require Contact: <uri>;text;duplex="half" m=audio 7200 RTP/AVP 0 98 a=rtpmap:98 t140c/8000 a=fmtp:98 cps=20 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 a=content:txp If the answering device is an ITD, with no specific indication that text would be used, it would accept only the G.711 audio. Contact: <uri>;text m=audio 7200 RTP/AVP 0 If the answering device is an ITD where the user has indicated a preference for text, it would accept the text/t140 medium and optionally G.711 audio. Contact: <uri>;text m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 Note that this response is only possible when the gateway had included text/140 in the offer. For other cases, the terminal needs to do a reinvite to add text to the call. D.2.1.2 Call from IP terminal A calling ITD could give a strong indication that it requires text support in the call by an INVITE according to this example: Contact: <sip:terminal@host.example.com>;text Accept-Contact: *;text;require m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 The network would make efforts to connect to a text capable gateway, that may acknowledge the text channel initially, because of the indication that text is required. The gateway should also start text channel connection procedures on the PSTN side. Contact: <gwy-uri>;text;duplex="half" m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 a=fmtp:99 cps=20 a=content:txp If on the other hand, the ITD user is not depending on text, but want to have it available for occasional use, the ITD can be configured to make its INVITE as follows: Contact: <sip:terminal@host.example.com>;text Accept-Contact: *;text m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 a=fmtp:99 The network would try to connect to a text capable gateway, but may connect through a non capable. The gateway may decide to not acknowledge the text channel initially, but indicates its text capability in the Contact header. Contact: <gwy-uri>;text;duplex="half" m=audio 7200 RTP/AVP 0 D.2.1.3 Text channel establishment during a call If at a later stage in the call, either party in the call starts text activity, and no text channel was set up initially, a text channel should be added to the session by a REINVITE. . This is an example of how it would be expressed from the gateway, Contact: <uri>;text;duplex="half" m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 a=fmtp:99 cps=20 a=content:txp The other party should answer with a similar sdp. If instead it is the ITD user who starts text activity, a text channel should be added to the session by a REINVITE from the ITD according to the following example. . Contact: <uri>;text m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 a=fmtp:99 cps=20 The gateway should answer with Contact: <uri>;text;duplex="half" m=audio 7200 RTP/AVP 0 m=text 7202 RTP/AVP 99 a=rtpmap:99 t140/1000 a=fmtp:99 cps=20 a=content:txp . The line "a=content:txp" is sent from the gateway as an indication to the ITD user that there is a likely need for the user to apply the formal turntaking habits that are common between PSTN text telephone users.. D.3 Procedure for terminals including both RFC 4103 and RFC 4351 support In the case when a terminal implements both RFC 4103 and RFC 4351 for text transport, the following procedures should be applied in the terminal. When establishing a call, an ITD may advertise support for ToIP in accordance with Annex B or Annex C, with one notable important exception: the ITD shall not include a list of supported modulations. Absence of the list of modulations shall be used as an indicator to the V.151 Gateway that the remote side is an ITD. Further, ITD devices shall not utilize SSEs in order to control state transition to ToIP. Rather, once a media flow is established to transport T.140 characters, either the Gateway or the ITD may transmit text immediately and without further negotiation. Payload type switching is used to transition between AUDIO and Text Relay modes. D.4 State transition and text handling The following description assumes that a V.151-compliant Gateway is in communication with an ITD and media flows have been established to transport text over IP in accordance with RFC 4103. When the gateway compliant to this annex detects a PSTN textphone that it supports using text relay, it shall autonomously connect to the textphone. After connection on the PSTN side, the gateway shall open the text channel to the ITD. It shall decode the characters received and transmit those to the ITD, giving proper respect to the maximum character-per-second (CPS) parameter advertised by the ITD. Determining the type of PSTN textphone device in use is the responsibility of the Gateway and the ITD need not concern itself with what kind of PSTN textphone device is connected to the Gateway. Likewise, when the ITD first has characters to send to the Gateway using text/140 payload type, the ITD opens the text channel if it was not opened before. Then the ITD transmits the characters to the Gateway. The Gateway shall perform a probing, as necessary, to determine the type of PSTN device connected, if any. While probing, the Gateway shall buffer any characters received and transmit those when probing completes and the gateway is connected to the text phone. The size of the character buffer is a matter of implementation, but should support the reception and buffering of characters for at least 60 seconds at the specified maximum character-per-second (CPS) value signaled to the ITD. If the Gateway does not support the modulation used by the PSTN textphone device, the Gateway may transmit the received textphone signals via VBD or the audio stream, depending on the capabilities of the ITD. Further, the Gateway may simply discard characters received from the ITD or transmit them to the PSTN textphone device using a pre-provisioned modulation. / OK Now? Gunnar ------------------------------------------------------------------------ ----- Gunnar Hellstrom, Omnitor gunnar.hellstrom@omnitor.se Mob: +46 708 204 288 Phone: +46 8 556 002 03 www.omnitor.se -----Original Message----- From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se] Sent: Friday, July 21, 2006 8:44 PM To: Arnoud van Wijk; 'Colin Perkins' Cc: mmusic@ietf.org Subject: RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-media-content-04.txt Colin, I am seaching in the draft amended V.151 Annex D for what is misleading you to believe that the IP terminal shall go half duplex. Can it be the proposed use of the duplex parameter from RFC 3840 Contact: <uri>;text;duplex="half" That may be where I have not used parameters properly. My intention is that this parameter value shall be used just to indicate the capability on the presentation level. On transport level between the IP terminal and the gateway the communication needs to be full duplex and not influenced by these parameters. What is your view on this use of the duplex parameter? One reason for the need to be full duplex between IP terminal and gateway is that there are more variations in simultaneous media handling of the far end PSTN terminal than we want to describe with parameters. It should rather be left for the users to adapt to the degree of simultaneity that is possible for the PSTN terminal we are connected to through the gateway. The V.21 textphones are full duplex in transport, but has varying handling of the presentation. Some merges the two sources in one window. Some have a kind of irc-like display with labels in front of the parties text. And yet some do a split in two windows of real-time text from each direction. There is no difference in call setup, so the gateway have no way to know the suitable level of simulaneity. And even if it is not suitable to write long stories from both ends simultanously when you have the one-window solution, you are very glad that there is a possibility to type $$$$$$ to ask for the floor while the other party is typing along. So, please do not ask us to complicate the IP implementation to try to adapt to the too limited and varying functionality of the PSTN textphones. Just warn the user, as can be done by using the discussed parameters. I have not before completely understood the intended use of the user-floor parameter. I thought it had something to do with control protocols for asking for the floor, but now I see that it is a close match to what we want to accomplish for text users. We want to tell the user that in this session you need likely to use the formal turntaking habits from PSTN text telephony, that is different from different country to country. Some use a * as turn indicator, some use a + and some use the letters GA. So, please let us have this way to simply give the user a coarse indication of the habits in the session. Do you want to delete also the user-floor parameter? Gunnar ------------------------------------------------------------------------ ---- - Gunnar Hellstrom, Omnitor gunnar.hellstrom@omnitor.se Mob: +46 708 204 288 Phone: +46 8 556 002 03 www.omnitor.se _______________________________________________ mmusic mailing list mmusic@ietf.org https://www1.ietf.org/mailman/listinfo/mmusic
- [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-media-c… Internet-Drafts
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Arnoud van Wijk
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Arnoud van Wijk
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- FW: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Jean-Francois Mule
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Desineni, Harikishan
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Arnoud van Wijk
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Frank Shearar
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Guido Gybels
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Even, Roni
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellstrom
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Guido Gybels
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellström
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Desineni, Harikishan
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellström
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Arnoud van Wijk
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Colin Perkins
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gonzalo Camarillo
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Arnoud van Wijk
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Cullen Jennings
- RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Gunnar Hellström
- Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-med… Paul Kyzivat
- [MMUSIC] Turntaking indication on the session lev… Gunnar Hellström
- [MMUSIC] Re: Turntaking indication on the session… Paul Kyzivat