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Re: [rtcweb] JSEP-01 Handling Early Media, using SIP, clarifications

Anne Raby <anne.raby@ericsson.com> Wed, 04 July 2012 14:13 UTC

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From: Anne Raby <anne.raby@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, Iñaki Baz Castillo <ibc@aliax.net>
Date: Wed, 04 Jul 2012 10:13:43 -0400
Thread-Topic: [rtcweb] JSEP-01 Handling Early Media, using SIP, clarifications
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP-01 Handling Early Media, using SIP, clarifications
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Thank you both for the clarifications.

/Anne
-----Original Message-----
From: Olle E. Johansson [mailto:oej@edvina.net] 
Sent: July-04-12 8:28 AM
To: Iñaki Baz Castillo
Cc: Anne Raby; rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-01 Handling Early Media, using SIP, clarifications


4 jul 2012 kl. 13:22 skrev Iñaki Baz Castillo:

> 2012/7/4 Olle E. Johansson <oej@edvina.net>:
>> Do we really need to map the PSTN call states into webrtc?
> 
> I don't suggest that. I was just explaining the SIP case in which
> different SDP are received by the UAC within the same INVITE
> transaction :)

And you where perfectly right. An OFFER can only have one ANSWER - but per early dialog in SIP. And one INVITE with an SDP offer can generate multiple early dialogs with multiple SDP answers, in PRACK or in 200 OK or somewhere else. The OFFER/ANSWER process is decoupled from the INVITE transaction, really.

/O ;-)

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