Re: [rtcweb] RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)]

Hadriel Kaplan <HKaplan@acmepacket.com> Tue, 20 September 2011 08:53 UTC

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [rtcweb] RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)]
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Date: Tue, 20 Sep 2011 08:56:11 +0000
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Cc: Randell Jesup <randell-ietf@jesup.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)]
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On Sep 20, 2011, at 4:48 AM, Olle E. Johansson wrote:

>> If it could, we'd probably have the siprec/remote-recording requirement accommodated.  :)
> Well, take a look at the source code for FreeSwitch or Asterisk and you'll see that "setting up a bridge" is not a piece of cake...
> You are making the assumption that you have no formatting issues and don't need to change framerate for video, orientation or anything else or audio transcoding.

Nope, I'm not making any such assumption.  I was just pointing out that if the browser did full mixing, then we'd have the remote recording use-case thing done too. 


>>> SIP has been very focused on device<->server interaction, not device<->device.  However:
>>> note that we have an app that knows why it has these calls in place; we're not defining an
>>> abstract, portable protocol use here.
>> 
>> Aha!  So it's not "SIP" that you meant... you meant "something that looks like SIP but isn't SIP per the RFCs".  ;)
>> 
> According to RFC 3261 SIP is a peer 2 peer protocol and not a device->server protocol. Just making a point.

I'm not sure if you meant that in response to my remark or Randell's?  What I meant by saying "it's not really SIP" wasn't because I think SIP is not device<->device (on the contrary, it is)... but rather that if rtcweb uses SIP but then doesn't actually follow the rules of SIP, then it's *not* SIP.

-hadriel