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:
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?
----
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
Klaghofer Karl ICN IB NL IP 7