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Re: [rtcweb] draft-sipdoc-rtcweb-open-wire-protocol-00 (Open In-The-Wire Protocol for RTC-Web)

Iñaki Baz Castillo <ibc@aliax.net> Mon, 31 October 2011 09:03 UTC

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Date: Mon, 31 Oct 2011 10:03:24 +0100
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From: Iñaki Baz Castillo <ibc@aliax.net>
To: Cullen Jennings <fluffy@cisco.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] draft-sipdoc-rtcweb-open-wire-protocol-00 (Open In-The-Wire Protocol for RTC-Web)
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2011/10/31 Cullen Jennings <fluffy@cisco.com>:
> One random idea … say that websockets was a protocol you could use to connect to a TURN server and then the TURN sever could send UPD or TCP SIP. That might be easier to deploy … not sure this is a good idea … just a random idea I thought I would mention.

There are two problems with this:

- If the websocket2tcp gateway is not SIP aware it would convert any
TCP packet into a WebSocket message. This means that such WebSocket
message could contain an incomplete SIP message or more than a single
SIP message, and that would require stream parsing in the WebSocket
client (JvaScript) becoming much more complex. In fact, our draft
mandates that a WebSocket message MUST contain a single and complete
SIP message (in the same way that SIP over UDP or SCTP). The draft
"XMPP over WebSocket" mandates the same.

- The SIP stack in JavaScript should set "TCP" in the Via transport,
or may be "TLS", or "UDP" (depending on the SIP transport used after
the websocket2tcp/udp/tls gateway.


So I strongly prefer to add WebSocket as a real SIP transport in a SIP
proxy (even more when I already have it deployed) :)

Regards.


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

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