Re: Questions on H.GCP applied on text telephony
John,
I like your idea of a script driven termination, and that without the script, the MGC gets all events and has to take all decisions.
Could be the appropriate model rather than the "user driven multiplexor" I described.
Do you then see the context on the IP side containing both streams for text and voice and that the modem side gets orders of what stream to feed at each moment. (Rather than changing the context back and forth to contain only the currently active IP-termination.)
Regards Gunnar
At 05:47 PM 1999-06-10 +0200, John Segers wrote:
Gunnar,
It took me a while to realize that you were addressing me with your question since the thread moved over from MEGACO to SG 16 list, but I clued in.
My idea about transitioning between text and audio was slightly different. I expected it would be driven by modem events. The modem detects a carrier, and in the absence of any scripts, notifies the controller. This then sends a command telling the Termination to change its media to text. A script in the MG could also specify this behavior, allowing us to get rid of the signaling to the MGC.
In the H.GCP document, I think there would have to be a description of the text medium as a subtype of data. And in the section explaining the modem parameters, there has to be a reference to V.18.
Does this sound reasonable?
Thank you for your input,
John Segers
Gunnar Hellstrom wrote:
The structure you described for bearers, terminations, multiplexors, contexts etc. described seems reasonable. For a PSTN termination in text telephone mode of the highest level, where the PSTN way of alternating between voice and text is supported, I see it best described as a kind of user-driven multiplexor.
When there is a carrier from the text phone, the multiplexor conveys the text stream.
When the carrier ceases, the multiplexor conveys the audio stream.
Text arriving from an IP termination may cause the state of the PSTN multiplexor to change from audio to text and then transmit the text in the prevailing text telephone mode.
Audio coming from an IP termination while the multiplexor is in the text mode can not go through. The multiplexor stays in the text mode until the user makes the alteration. ( usually by lifting the handset and "stealing" the line from the textphone.)
On the IP side, H.323 Annex G is supposed to support simultaneous audio and text. There is no need for this kind of user driven multiplexor there.
I understand that the requirements behind this description should go into section 11.2.4 of megaco requirements (D-286 of Santiago). But where in H.gcp do you want the description of parameters and functions needed in the gateway components for this functionality?
Gunnar Hellström
At 10:09 AM 1999-06-10 +0200, John Segers wrote:
Hao,
In the current MEGACO model as described in draft-01, a Termination describes the media and in-band signaling for one user, and how they are transported etc. A Context associates all Terminations in a session and implies audio mixing if there are more than two.
This idea is retained in the Santiago work. Going to multiple media, a Termination still describes the media and in-band signaling for one user. A Context still associates all Terminations in a session. For multimedia sessions, the mixing/switching can no longer be implied, but will be described by means of Context properties. If there is only audio, nothing changes in the Context.
In order to allow H.320 and H.324 where multiple media are multiplexed on one or more bearers, there needs to be a distinction between TerminationID and BearerID in order to retain the concept that a Termination describes the media, in-band signaling and transport.
You see, the goal was to stay close to the basic monomedia connection model for multimedia, always keeping in mind that there should be as little overhead as possible for controlling voice calls (setup, teardown, ...). The places where you see overhead occurring in the Santiago proposal are indeed when dealing with multiple bearers for multimedia, and, as Brian Rosen pointed out, in Audits. It seems to me that this is acceptable because these operations will occur much less frequently than the voice/fax/NAS setups.
Regards,
John Segers
HaoHou wrote:
Jonh,
Thanks for your reply,
Please see comments below.
Second, the use of Add/Subtract/Modify commands. You are proposing to use Add/Subtract to add media streams to a session or delete them from it. This changes the semantics of the commands drastically. They are intended to perform operations on Contexts (add or remove Terminations). In the Santiago proposal this semantics is maintained. Operations to change Termination parameters are performed using the Modify command.
Regards,
John Segers
Thats why the Santiago proposal makes the Modify command awkward. ? Suppose we have 3 bearer channels already in a Termination. In order to release one bearer channel, we have to list both the
other two
bearer channels in the MODIFY command. In order to add one more bearer channel, we have to list the new bearer channel and all the existed bearer channels in the MODIFY command.
I dont think the change of semantics will affect the protocol and the implementation too much. Instead, it will make the protocol easier to be understood: As long as you want to add some entities into the context or
termination, you
use ADD. As long as you want to sbustract some entities from the context or termination, you use Subtract. It is a more natural way to express
the
intention of the commands.
Could you tell me why Santiago proposal wants to keep the old semantics unchanged, after the big change of the Context and Termination concept? What does
it gain?
Id rather change the semantics to meet the change of the concept.
Best Regards,
Hao
-- John Segers email: jsegers@lucent.com Lucent Technologies Room HE 306 Dept. Forward Looking Work phone: +31 35 687 4724 P.O. Box 18, 1270 AA Huizen fax: +31 35 687 5954
Gunnar Hellström
Ericsson Home Systems
Tel +46 751 100 501 Fax +46 8 556 002 06
e-mail gunnar.hellstrom@omnitor.se Video +46 8 556 002 05
-- John Segers email: jsegers@lucent.com Lucent Technologies Room HE 306 Dept. Forward Looking Work phone: +31 35 687 4724 P.O. Box 18, 1270 AA Huizen fax: +31 35 687 5954
------------------------------------- Gunnar Hellström
Ericsson Home Systems
Tel +46 751 100 501 Fax +46 8 556 002 06
e-mail gunnar.hellstrom@omnitor.se Video +46 8 556 002 05 -------------------------------------
Gunnar,
I'm glad you like the idea. My intention was to have some discussion with you and the H.GCP editor, and then post the outcome of that to the list(s) to see what people think of it. But I guess we might as well have our discussions on the list.
Your question was what to do on the IP side: should both streams, text and voice, be there continuously or should they be switched. My feeling is that they should both be there all the time. If the DS0 side has a script attached to perform the switching without intervention of the MGC, it would be more convenient to have the IP side ready to receive both streams. Otherwise a change on the DS0 side (V.18 detected) would have to result in operations on Context and IP side Termination. This changes the idea that scripts are attached to Terminations, just like digit maps.
I'd like to have the opinion of people on the MEGACO list as well, hence the cross post.
Regards,
John Segers
Gunnar Hellstrom wrote:
John,
I like your idea of a script driven termination, and that without the script, the MGC gets all events and has to take all decisions.
Could be the appropriate model rather than the "user driven multiplexor" I described.
Do you then see the context on the IP side containing both streams for text and voice and that the modem side gets orders of what stream to feed at each moment. (Rather than changing the context back and forth to contain only the currently active IP-termination.)
Regards Gunnar
At 05:47 PM 1999-06-10 +0200, John Segers wrote:
Gunnar,
It took me a while to realize that you were addressing me with your question since the thread moved over from MEGACO to SG 16 list, but I clued in.
My idea about transitioning between text and audio was slightly different. I expected it would be driven by modem events. The modem detects a carrier, and in the absence of any scripts, notifies the controller. This then sends a command telling the Termination to change its media to text. A script in the MG could also specify this behavior, allowing us to get rid of the signaling to the MGC.
In the H.GCP document, I think there would have to be a description of the text medium as a subtype of data. And in the section explaining the modem parameters, there has to be a reference to V.18.
Does this sound reasonable?
Thank you for your input,
John Segers
Gunnar Hellstrom wrote:
The structure you described for bearers, terminations, multiplexors, contexts etc. described seems reasonable. For a PSTN termination in text telephone mode of the highest level, where the PSTN way of alternating between voice and text is supported, I see it best described as a kind of user-driven multiplexor.
When there is a carrier from the text phone, the multiplexor conveys the text stream.
When the carrier ceases, the multiplexor conveys the audio stream.
Text arriving from an IP termination may cause the state of the PSTN multiplexor to change from audio to text and then transmit the text in the prevailing text telephone mode.
Audio coming from an IP termination while the multiplexor is in the text mode can not go through. The multiplexor stays in the text mode until the user makes the alteration. ( usually by lifting the handset and "stealing" the line from the textphone.)
On the IP side, H.323 Annex G is supposed to support simultaneous audio and text. There is no need for this kind of user driven multiplexor there.
I understand that the requirements behind this description should go into section 11.2.4 of megaco requirements (D-286 of Santiago). But where in H.gcp do you want the description of parameters and functions needed in the gateway components for this functionality?
Gunnar Hellström
-- John Segers email: jsegers@lucent.com Lucent Technologies Room HE 306 Dept. Forward Looking Work phone: +31 35 687 4724 P.O. Box 18, 1270 AA Huizen fax: +31 35 687 5954
participants (2)
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Gunnar Hellstrom -
John Segers