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Re: [rtcweb] Support for websockets

Matthew Kaufman <matthew.kaufman@skype.net> Mon, 05 September 2011 03:11 UTC

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Date: Sun, 04 Sep 2011 20:12:57 -0700
From: Matthew Kaufman <matthew.kaufman@skype.net>
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To: Harald Alvestrand <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Support for websockets
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On 7/26/2011 5:50 AM, Harald Alvestrand wrote:
> On 07/26/11 01:28, Avasarala, Ranjit wrote:
>> Hi all
>>
>> By when can we expect WebRTC code (Google code) to support Websockets?
> ...
> In this forum, it may be appropriate to describe exactly how you see 
> Websockets fitting in with the RTCWEB specification suite; so far, 
> it's seemed to be not very closely related.

I think there's a legitimate question as to how transmission of media 
over TCP should work. I believe that the existing code bases all assume 
that if you get IP addresses, you use them; if you get IP addresses for 
TURN servers you talk to them over UDP and use TURN for relaying; and if 
you get IP addresses for TURNS servers you talk to them over TLS/TCP and 
use them for relaying. Therefore the only TCP transport is 
TURN-over-TLS-over-TCP.

Is that sufficient and reasonable, or should media-over-websockets (or 
something else) be how TCP transmission of media works?

Matthew Kaufman

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