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

Iñaki Baz Castillo <ibc@aliax.net> Mon, 26 September 2011 17:48 UTC

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Date: Mon, 26 Sep 2011 19:50:47 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Jozsef Vass <jovass@adobe.com>
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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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2011/9/26 Jozsef Vass <jovass@adobe.com>;:
> Look at it from a game developer's perspective. I do not want to build a softphone phone with advanced features, just to add voice chat to the game. Online registered players should be able to chat with each other, but really no need to interoperate with anything else (and preferred not to).  Quite often, it is a group of people who are playing together, so multiparty chat is highly desirable.  As I game developer, I do not event know what is SIP, Jingle or SDP and probably do not even care about it.

Indeed there should be a simple mechanism for allowing web developers
to run audio/video sessions in their web applications, but that should
not avoid or make impossible for others to run a SIP or XMPP/Jingle
client within a web.

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