Re: [rtcweb] "20 lines" (Re: RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)])

Jozsef Vass <jovass@adobe.com> Mon, 26 September 2011 17:17 UTC

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From: Jozsef Vass <jovass@adobe.com>
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Date: Mon, 26 Sep 2011 10:19:17 -0700
Thread-Topic: [rtcweb] "20 lines" (Re: RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)])
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] "20 lines" (Re: RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)])
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All these advanced features will probably be used by a handful of companies/developers. I would rather see something simple, xml based that is more compatible with the web than SDP. 

Jozsef

-----Original Message-----
From: Iñaki Baz Castillo [mailto:ibc@aliax.net] 
Sent: Monday, September 26, 2011 12:55 AM
To: Jozsef Vass
Cc: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] "20 lines" (Re: RTCWeb default signaling protocol [was RE: About defining a signaling protocol for WebRTC (or not)])

2011/9/23 Jozsef Vass <jovass@adobe.com>;:
> there is no need for interoperability, there is no need for SIP, Jingle, etc. At the minimum, this would require a simple service for ip address lookup and to aid NAT and firewall traversal.
> For example, when a user registers with this service, he/she is assigned an identifier or blob. In order to communicate with another party, all they need to do is to exchange this information (this can be done using websocket, etc.) and setup media channel. There should be no need for parsing SDP, etc.


...and when you need more features (like adding video within an existing audio session, or putting the session on hold, or negotiating codecs and encryption parameters) you will realize that you are reinventing SDP.





--
Iñaki Baz Castillo
<ibc@aliax.net>;