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

["Olle E. Johansson" <[e-mail]>]

20 sep 2011 kl. 10:17 skrev Hadriel Kaplan:

> 
> On Sep 20, 2011, at 3:45 AM, Randell Jesup wrote:
> 
>> Ok; I hadn't looked at what controls we're giving the JS app (mostly focusing on IETF
>> level stuff).  W3C issue.  It would be nice if an app could set up a bridge; I'm
>> a little surprised it can't.
> 
> 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.

> 
> 
>>> A full mesh is what *should* happen, but SIP/SDP can't do it, afaict.  It would treat them either as independent calls even at a media layer, or as a full-mixer conference focus model.  The closest thing we have would probably be the Join header, but I believe it's semantics is to join as a full mixer conf call.  Isn't this full-mesh media-forking thing actually a new semantic for SIP/SDP?  (it's hard to believe with 100+ drafts/RFCs this scenario hasn't already been addressed in SIP - I must be just having a memory leak)
>> 
>> 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.

/O