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Re: [rtcweb] RTCWeb Forking usecase [was RE:draft-kaplan-rtcweb-sip-interworking-requirements-00]

"Ravindran Parthasarathi" <pravindran@sonusnet.com> Fri, 28 October 2011 20:13 UTC

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Thread-Topic: [rtcweb] RTCWeb Forking usecase [was RE:draft-kaplan-rtcweb-sip-interworking-requirements-00]
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From: Ravindran Parthasarathi <pravindran@sonusnet.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Iñaki Baz Castillo <ibc@aliax.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWeb Forking usecase [was RE:draft-kaplan-rtcweb-sip-interworking-requirements-00]
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Hi Christer,

In case of SIP forking interop with WebRTC, it is possible for WebRTC-SIP gateway to handle this scenario and not by browser. In fact, it is my review comment to draft-kaplan-rtcweb-sip-interworking-requirements-00 (Para 2 in http://www.ietf.org/mail-archive/web/rtcweb/current/msg02316.html) where this mail thread started.

Even if WebRTC supports both serial & parallel forking, whether WebRTC signaling will take up the same shape of SIP forking wherein initial INVITE only forks and not for RE-INVITE? In case of WebRTC forking, it is very much possible for different shape of forking. The well defined WebRTC usecase & requirement may clarify these doubts.

Thanks
Partha




>-----Original Message-----
>From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>Sent: Saturday, October 29, 2011 1:20 AM
>To: Ravindran Parthasarathi; Iñaki Baz Castillo
>Cc: rtcweb@ietf.org
>Subject: RE: [rtcweb] RTCWeb Forking usecase [was RE:draft-kaplan-
>rtcweb-sip-interworking-requirements-00]
>
>
>Hi,
>
>In my opinion, interworking with SIP is itself such use-case :)
>
>I think the question is whether a browser should be able to handle
>parallel forking, or if serial forking is enough.
>
>In my opinion serial forking is enough, ie the JS app (or, if you want,
>a gateway) informs the browser about the currently active remote media
>parameters.
>
>Regards,
>
>Christer
>
>
>________________________________________
>From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of
>Ravindran Parthasarathi [pravindran@sonusnet.com]
>Sent: Friday, October 28, 2011 10:43 PM
>To: Iñaki Baz Castillo
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] RTCWeb Forking usecase [was RE:   draft-kaplan-
>rtcweb-sip-interworking-requirements-00]
>
>Inaki,
>
>ISTM, SIP specific functionality is pushed into WebRTC client
>unnecessary without well-defined WebRTC usecase.
>
>Thanks
>Partha
>
>>-----Original Message-----
>>From: Iñaki Baz Castillo [mailto:ibc@aliax.net]
>>Sent: Saturday, October 29, 2011 1:04 AM
>>To: Ravindran Parthasarathi
>>Cc: Harald Alvestrand; rtcweb@ietf.org; Hadriel Kaplan
>>Subject: Re: [rtcweb] RTCWeb Forking usecase [was RE: draft-kaplan-
>>rtcweb-sip-interworking-requirements-00]
>>
>>2011/10/28 Ravindran Parthasarathi <pravindran@sonusnet.com>:
>>> As I mentioned earlier, SIP (serial) forking requirement shall be met
>>by gateway signaling and no extra requirement for browser.
>>
>>Do it in your gateway if you want, but don't try to mandate it please.
>>As I also mentioned earlier, some of us are willing to handle forking
>>in the WebRTC client (at JavaScript level) and that is perfectly
>>possible.
>>
>>Regards.
>>
>>--
>>Iñaki Baz Castillo
>><ibc@aliax.net>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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