Dear multimedia experts.

In my efforts to establish the simple IP voice and text telephone Text SET, I came across a section in H.323 Annex F (Simple Endpoint Type, TD11 in Monterey) that I feel is causing a functional obstacle also to the voice users. Can anyone clarify if I am correct and why it is specified the way it is.

In section 7.6.1 and 7.6.2 it is specified:" The Audio SET device shall then resume transmitting its media stream(s) to the transport address(es) newly indicated in the OpenLogicalChannel structures." I understand that this means that you cannot re-negotiate audio coding, and you cannot add text conversation after rerouting the call from a Voice only SET to a Text SET.

Re-negotiating the audio coding will probably be a desired function, e.g. when rerouting from a fixed to a wireless IP phone. Adding a data channel for text will also be a desired function, after answering a call in an audio-only SET, and then rerouting it to a text-capable SET. That action is very common in today's text telephone usage, and I would expect it to be just as common in the IP telephony world. You first receive the call in the terminal that is closest to you, and then you get a reason to start text mode. Then you transfer the call to another device with text capabilities, where you can switch mode.

Questions:

1. Is that kind of call transfer that is handled by the mechanisms in 7.61 and 7.6.2?

2. Are my conclusion right about the limitations?

3. Is this limitation a consequence of using Fast Connect?

4. Do you see any possibility to avoid the negative effects of it - to make re-negotiation possible?

5. Is the specified functionality acceptable in the voice world? If two devices have agreed on a voice coder, is it likely that the third device supports it? Will this not create a lot of unsuccessful call transfers where the users will have a no chance to understand why they fail?

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Another question area:

6. When selecting the transport protocol for the text conversation, the current draft (APC 1504) specifies TCP or UDP. I realize that there are situations where TCP must be avoided. One such situation is a sub-titled H.332 transmission. Also other multi-casting situations is better off with a UDP based transport protocol. I am therfore now leaning towards using RTP as the transport for text conversation. With RTP we can discover dropped frames and possibly invent a mechanism to mark that event in the text stream for T.140 to display. If we have less than 3 % dropped frames, I think the users would accept it.

6.1 Do you agree that there are situations when TCP should be avoided, and a UDP based protocol preferred?

6.2 Do you agree that RTP is a good alternative, with a thin protocol for error indications to the user?

6.3 Most packets will carry only 1-4 characters . Can anyone give me an indication on the expected packet loss rates in different situations for such packets. Or a document giving such figures. Is max 3% loss reachable?

Please give your view on these questions.

Best regards

Gunnar Hellström ----------------------------------------------- Gunnar Hellstrom Representing Ericsson in ITU-T

E-mail gunnar.hellstrom@omnitor.se Tel +46 751 100 501 fax +46 8 556 002 06