basic ASN.1 coding question

henri.maenpaa at NOKIA.COM henri.maenpaa at NOKIA.COM
Mon Mar 22 04:49:15 EST 1999


Frank, Karl, Anyone else who's interested,

A limitation in H.450.3 (Call Forward) recently became clear to me, and the
solution may arrive through the sorts of things Frank's suggesting.

Background:
What I see as a problem arises from the endpoint-centric nature of CFU (Call
Forward Unconditional), especially in the case where the endpoint being
forwarded is PC-based (probably NOT a SET).

The specific problem:
I would regard as a prime use for CFU to be the ability to set up forwarding
and then switch off my PC, so that all my calls would then go (for example)
to a colleague.  Everything seems to be there in H.450.3 to allow this to
happen in the gatekeeper-routed model (obviously the GK-routed model is
likely to be required, because the GK appears to be the only "live" entity
that can do the forwarding), with the exception of the transaction between
my PC and its gatekeeper.

The solution:
I don't know.  Maybe an H.450.3 extension (which has dangers because a GK is
not normally a callable entity), maybe a new RAS transaction type (but we
don't want proliferation of RAS messages)?

The plea:
I'd like to see some serious suggestions on this, and really drive some
solution towards standardisation, either as an Annex to H.450.3 or
elsewhere.  It seems to me that this is possibly the most important
application of Call Forwarding, so it seems a shame that it is missing.

If consensus works towards a RAS-based solution, we may be able to do this
in a way that will also suit Frank's ideas?

Regards,
Chris

> -----Original Message-----
> From: Derks, Frank [mailto:F.Derks at PBC.BE.PHILIPS.COM]
> Sent: 22 March 1999 10:43
> To: ITU-SG16 at MAILBAG.INTEL.COM
> Subject: Re: AW: Call hold and transfer in H.323 AnnexF. Too limited??
>
>
> Folks,
>
> when talking about a Simple Endpoint Type, I think we should
> aim for it to
> be something that closely resembles a black phone. This way
> it becomes a lot
> easier to define what its capabilities are and it makes life
> easy on the
> users and on those companies that will actually make
> (physical) IP-phones.
> These phones should probably look and act like the normal
> phones that are
> currently being used. Looking at how most supplementary services are
> accessed in both the public and the private (PBX) networks, I
> think it is
> safe to say that in most cases we are talking about "stimulus
> protocols".
> I.e. DTMF digits are sent to an exchange and the exchange
> interprets certain
> digit sequences as being the invocation of some service
> rather than a number
> to be dialled. The big advantage over functional protocols
> (like H.450.x)
> being that services can be added from the exchange side, without the
> terminal having to be modified as well.
>
> Functional protocols never became a success in  the ISDN
> world and this may
> end up to be the same in the IP world. However, having said
> this, there is a
> lot more potential for easy upgrading of e.g. terminal
> software in this
> domain, which reduces the side effects of functional protocols.
>
> It does not seem to make sense to define "alternative"
> mechanisms to provide
> services, so I would strongly opt for using H.450.x when
> possible and using
> a simple stimulus protocol otherwise. The latter would allow service
> providers to easily make services available and I see no
> reason why this
> should be standardised. In practice, today, we already see
> that certain
> digit sequences for service activation are identical in
> several countries.
>
> Frank
>
> -----------------------------------------------------
> Frank Derks                    |Tel  +31 35 6893238 |
> Advanced Development           |Fax  +31 35 6891030 |
> Philip Business Communications |P.O. Box 32         |
>                                |1200 JD  Hilversum  |
>                                |The Netherlands     |
> ----------------------------------------------------|
> E-mail: mailto:f.derks at pbc.be.philips.com           |
> WWW: http://www.business-comms.be.philips.com       |
> -----------------------------------------------------
>
>
>
> -----Original Message-----
> From: Klaghofer Karl ICN IB NL IP 7
> [mailto:Karl.Klaghofer at ICN.SIEMENS.DE]
> Sent: 18 March 1999 22:36
> To: ITU-SG16 at MAILBAG.INTEL.COM
> Subject: 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 at TIES.ITU.INT]
> > Gesendet am:  Donnerstag, 18. März 1999 18:57
> > An:   ITU-SG16 at 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 at ICN.SIEMENS.DE>
> > To: ITU-SG16 at MAILBAG.INTEL.COM <ITU-SG16 at 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 at icn.siemens.de
> > >
> > >
> > >
> > >> -----Ursprüngliche Nachricht-----
> > >> Von:  Gunnar Hellstrom [SMTP:gunnar.hellstrom at OMNITOR.SE]
> > >> Gesendet am:  Dienstag, 16. März 1999 23:01
> > >> An:   ITU-SG16 at 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 at omnitor.se
> > >> Tel +46 751 100 501
> > >> fax +46 8 556 002 06
>



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