Re: [rtcweb] Agenda requests for Atlanta meeting

Cullen Jennings <fluffy@iii.ca> Tue, 09 October 2012 01:41 UTC

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From: Cullen Jennings <fluffy@iii.ca>
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Date: Mon, 8 Oct 2012 18:41:34 -0700
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To: Roman Shpount <roman@telurix.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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On Oct 8, 2012, at 18:31 , Roman Shpount <roman@telurix.com> wrote:

> On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> 
> Well, what's the end user experience you're trying to achieve  - lets say we are talking about audio. Do you want to play the first one or the last one? If you want to bridge them into a party line, this is probably not how you want the SIP side to happen.
> 
> 
> This is still serial forking. Imagine the following scenario: You are calling a phone number, you dial first provider who starts playing a dial tone then uses an UPDATE to connect you to a media server to play an announcement after which your call goes to voice mail (another dialog and another SDP). This is all fairly legal and not unusual for SIP scenarios. Nothing to do with party lines.
> _____________
> Roman Shpount
> 

Ok, serial forking with early media seems like it should work. Seems like the browser would make the offer, that gateway turned into SIP invite that then went to sequential forking proxy that send sent the invite to UA A. UA A returns a 180 that gets mapped to PRANSWER, the proxy then forks to UA B that sends a 200 on a different dialog that gets mapped to answer.