Re: dtmf relay in SIP?

"Krishna Sai" <ksai@bigfoot.com> Tue, 03 August 1999 14:02 UTC

Return-Path: <owner-confctrl>
Received: (from majordom@localhost) by zephyr.isi.edu (8.8.7/8.8.6) id HAA27304 for confctrl-outgoing; Tue, 3 Aug 1999 07:02:13 -0700 (PDT)
Received: from tnt.isi.edu (tnt.isi.edu [128.9.128.128]) by zephyr.isi.edu (8.8.7/8.8.6) with ESMTP id HAA27299 for <confctrl@zephyr.isi.edu>; Tue, 3 Aug 1999 07:02:11 -0700 (PDT)
Received: from usmlns02.jax.ecitele.com ([12.27.128.28]) by tnt.isi.edu (8.8.7/8.8.6) with SMTP id HAA24123 for <confctrl@ISI.EDU>; Tue, 3 Aug 1999 07:02:10 -0700 (PDT)
Received: from ksai ([172.20.11.96]) by usmlns02.jax.ecitele.com (Lotus SMTP MTA v4.6.5 (863.2 5-20-1999)) with SMTP id 852567C2.004D0A30; Tue, 3 Aug 1999 10:01:28 -0400
Message-ID: <001a01beddb8$aec35370$600b14ac@ksai.ecijax.com>
Reply-To: Krishna Sai <ksai@bigfoot.com>
From: Krishna Sai <ksai@bigfoot.com>
To: Jacob Avraham <jacoba@mediagate.co.il>, confctrl@ISI.EDU
Subject: Re: dtmf relay in SIP?
Date: Tue, 03 Aug 1999 10:01:28 -0400
MIME-Version: 1.0
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: 7bit
X-Priority: 3
X-MSMail-Priority: Normal
X-Mailer: Microsoft Outlook Express 4.72.3110.5
X-MimeOLE: Produced By Microsoft MimeOLE V4.72.3110.3
Sender: owner-confctrl@zephyr.isi.edu
Precedence: bulk

The avt (audio/video transport) group works on these issues. In particular,
refer to drafts like :
http://www.ietf.org/internet-drafts/draft-ietf-avt-dtmf-01.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-tones-00.txt

regards,
Krishna Sai
ECI Telecom

-----Original Message-----
From: Jacob Avraham <jacoba@mediagate.co.il>
To: confctrl@ISI.EDU <confctrl@ISI.EDU>
Date: Tuesday, August 03, 1999 5:44 AM
Subject: RE: dtmf relay in SIP?


>
>
>
>I'm curious about DTMF in-band within the RTP stream. My audio guys tell me
>that once the audio is encoded with something like G.723 or some other
>vocoder, you loose the tone frequency info and it's almost impossible
>to get accurate detection of DTMF.
>Possibly, one could change RTP payload mid-stream to something like G.711
>but that's not always feasible (and forbidden by other protocols like
>H323).
>Any real world experience with this issue?
>
>Thanks,
>
>Jacob Avraham
>Mediagate
>>
>> This configuration is a classic case of where RTP encapsulation is best.
>> I'm personally unconvinced about dtmf transport in SIP in general; here
>> its definitely not needed.
>>
>> Thanks,
>> Jonathan R.
>>
>> gonzalo.camarillo@lmf.ericsson.se wrote:
>> >
>> > Hi,
>> >
>> > > If I have a regular telephone connected to a SIP enabled gateway and
>calls
>> > > another telephone with similar connection, is there anyway I can do
>dtmf
>> > > relay between the two phones in the SIP network?  That is, when the
>gateway
>> > > receives a dtmf tone from the connected telephone, how does the
>gateway send
>> > > out the dtmf tone in a SIP message?
>> >
>> > You can use either the SIP INFO method or just encode the DTMF tones in
>RTP
>> > packets...
>> >
>> > Regards,
>> >
>> > Gonzalo
>
>