AW: Call hold and transfer in H.323 AnnexF. Too limited??

Paul E. Jones paul.jones at
Thu Mar 18 12:56:57 EST 1999


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.


-----Original Message-----
From: Klaghofer Karl ICN IB NL IP 7 <Karl.Klaghofer at ICN.SIEMENS.DE>
Date: Wednesday, March 17, 1999 3:27 PM
Subject: AW: Call hold and transfer in H.323 AnnexF. Too limited??

>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
>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
>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
>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
>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 at
>> -----Ursprüngliche Nachricht-----
>> Von:  Gunnar Hellstrom [SMTP:gunnar.hellstrom at OMNITOR.SE]
>> Gesendet am:  Dienstag, 16. März 1999 23:01
>> An:   ITU-SG16 at
>> 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
>> to start text mode. Then you transfer the call to another device with
>> capabilities, where you can switch mode.
>> Questions:
>> 1. Is that kind of call transfer that is handled by the mechanisms in
>> 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
>> 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,
>> 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
>> Please give your view on these questions.
>> Best regards
>> Gunnar Hellström
>> -----------------------------------------------
>> Gunnar Hellstrom
>> Representing Ericsson in ITU-T
>> E-mail gunnar.hellstrom at
>> Tel +46 751 100 501
>> fax +46 8 556 002 06

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