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Re: [rtcweb] JSEP: Order of m- lines in multiple PeerConnections

Christer Holmberg <christer.holmberg@ericsson.com> Fri, 18 October 2013 19:35 UTC

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] JSEP: Order of m- lines in multiple PeerConnections
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Date: Fri, 18 Oct 2013 19:34:45 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP: Order of m- lines in multiple PeerConnections
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Hi,

Again, it is not specific to parallel forking - it is also needed for SERIAL forking, in cases where you need to send updated Offers on early dialogs, and therefore cannot use PRANSWER.

As far as the solution is concerned: on a high level, whenever you create a new PC (or, when you add the media streams to it), you need to be able to indicate the order of the m- lines. Whether it is done explicitly, or implicitly e.g. by somehow referencing the old PC, I don't know, but I don't think it should be very difficult.

Regards,

Christer

-----Alkuperäinen viesti-----
Lähettäjä: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com] 
Lähetetty: 18. lokakuuta 2013 22:12
Vastaanottaja: Christer Holmberg
Kopio: Suhas Nandakumar; rtcweb@ietf.org
Aihe: Re: [rtcweb] JSEP: Order of m- lines in multiple PeerConnections


On Oct 17, 2013, at 10:59 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:

> And, I'm not asking for a solution at this point, simply that we identify it as an issue that needs to be solved :)
> 
> 

So I sort of disagree on two points here. 

I disagree that it needs to be solved - I'm not against solving it if anyone has an easy way but every time we talk about this the conclusions comes up people don't want to bother to fully solve the parallel forking problem in the first version of the webrtc. Speaking purely for myself, SIP parallel forking has not turned out to be extremely useful and has turned out to seriously complicate the use and extensions to the protocol - basically the HERFP problem - so I don't really care if webrtc takes on that problem or not. 

Second, I suspect that Invite with replaces actually does solve this. 

I certainly don't mind marking it as an issue but it's not clear to me that the WG thinks it is an issue that needs to be solved or that it an issue that is not solved. I've sort of been waiting to see the other "easy" stuff in JSEP / Bundle / Unified plan get sorted out and then figured we could go back and see what was possible or not with parallel forking in SIP.

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