AW: AW: Call hold and transfer in H.323 AnnexF. Too limited??
See comment below.
Karl
-----Ursprüngliche Nachricht----- Von: Paul E. Jones [SMTP:paul.jones@TIES.ITU.INT] Gesendet am: Donnerstag, 18. März 1999 18:57 An: ITU-SG16@mailbag.cps.intel.com Betreff: Re: AW: Call hold and transfer in H.323 AnnexF. Too limited??
Karl,
Unfortunately, I will have to disagree with your comments. While it is true that the H.450 supplementary services could be utilized in a SET device, I believe that introducing H.450 into a SET breaks the spirit of that work.
The goal of Annex F is to define a "Simple Endpoint Type". There are simpler ways to put a call on hold and to transfer a call. Introducing H.450 introduces a lot more complexity that I believe we want to have. If Annex F is not sufficiently clear on how to simply transfer a call or put a call on hold, we should work on that text-- I will absolutely disagree with introducing H.450 into a SET device.
[Klaghofer, Karl PN VS LP3] Whatever you mean with "introducing" - H.450 as I sayd in my previous mail is a way of providing supplementary services like call hold and call transfer to a SET device. It IS already part of the H.323 Annex F!
Paul
-----Original Message----- From: Klaghofer Karl ICN IB NL IP 7 Karl.Klaghofer@ICN.SIEMENS.DE To: ITU-SG16@MAILBAG.INTEL.COM ITU-SG16@MAILBAG.INTEL.COM Date: Wednesday, March 17, 1999 3:27 PM Subject: AW: Call hold and transfer in H.323 AnnexF. Too limited??
Gunnar,
You are referring to call hold and transfer in conjunction with H.323
Annex
F SETs (Audio or Text) and clause 7.6 of H.323 Annex F.
Talking about call hold, clause 7.6 of H.323 Annex F is not needed for a
SET
at all. Call Hold works for a SET as it is defined in H.450.4.
Talking about Call Transfer, clause 7.6 of H.323 Annex F is not needed
for a
SET, if the transfer is executed by the endpoints as defined in H.450.2. Codec re-negotiation you are referring to is no problem and takes place between the transferred and the transferred-to endpoint. This may cover
your
case with wireless endpoints being involved.
For call transfer, section 7.6 of H.323 Annex F is only needed if the gatekeeper or a proxy acts on behalf of the transferred SET endpoint B. However, media re-negotiation also should work here as part of the
fastStart
method.
Regards, Karl
Karl Klaghofer, Siemens AG, Dpmt. ICN IB NL IP7 Hofmannstr. 51, D-81359 Munich, Germany Tel.: +49 89 722 31488, Fax.: +49 89 722 37629 e-mail: karl.klaghofer@icn.siemens.de
-----Ursprüngliche Nachricht----- Von: Gunnar Hellstrom [SMTP:gunnar.hellstrom@OMNITOR.SE] Gesendet am: Dienstag, 16. März 1999 23:01 An: ITU-SG16@mailbag.cps.intel.com Betreff: Call hold and transfer in H.323 AnnexF. Too limited??
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:
- Is that kind of call transfer that is handled by the mechanisms in
7.61
and 7.6.2?
Are my conclusion right about the limitations?
Is this limitation a consequence of using Fast Connect?
Do you see any possibility to avoid the negative effects of it - to
make re-negotiation possible?
- 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?
Another question area:
- 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
participants (1)
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Klaghofer Karl ICN IB NL IP 7