Re: [rtcweb] SIP vs Websocket in RTCWeb [was RE: SIP MUST NOT be used in browser?]

Dzonatas Sol <> Mon, 12 September 2011 17:10 UTC

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Date: Mon, 12 Sep 2011 10:14:22 -0700
From: Dzonatas Sol <>
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Subject: Re: [rtcweb] SIP vs Websocket in RTCWeb [was RE: SIP MUST NOT be used in browser?]
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On 09/12/2011 08:55 AM, Peter Saint-Andre wrote:
> On 9/12/11 12:57 AM, Ravindran Parthasarathi wrote:
>> Changing the title for giving the clear context of discussion.
>> Peter,
>> Thanks for forwarding info about draft-ietf-hybi-thewebsocketprotocol. I
>> started this mail thread to know whether RTCWeb1.0 is a unofficial
>> RFC3261bis for the line side (endpoint to access server) :-) [I really
>> don't know the better term for the line side]. Endpoint may be desktop,
>> smart phone (android), laptop, tablet, CPE, etc.,
>> Till reading this draft, I assumed websocket as a socket layer for HTTP
>> and it is bad assumption :-(. In short, browser is able to create two
>> way communication with webserver (which has globally routable address).
>> Two browser creating websocket with web servers will be able to
>> communicate with each other. This architecture exactly fits in SIP world
>> as
>>                         SIP UA<---->B2BUA<----->SIP UA
>> And resultant as   browser<--->  webserver<---->  browser. I tend to
>> agree with you that Websocket looks as a better choice for this simple
>> web architecture as there is no need of identity exchange here because
>> webserver knows and authenticated both browsers with the corresponding
>> identity. In fact, B2BUA with globally routable address will interop
>> better with any endpoint for that matter. The difference comes into
>> picture for federation (interop between servers). I'm not very clear
>> whether websocket is intended for federation as well or not. Most of the
>> discussion RTCWeb points to use SIP as a federation protocol which may
>> change later. I'm interested knowing your view here. For this mail, I
>> assume that SIP as a federation protocol of RTCWeb1.0 and I'm ready to
>> change if it is the right thing to do :-)
> I think that the protocol used for server-to-server federation is a
> matter for the service providers and thus is not in scope for RTCWeb.
> Some s2s links might use SIP, some might use XMPP/Jingle, etc.
> Peter

Yesterday I thought browser-hints could help with transitions to 
federation, especially for RTCWeb-to-SIP:


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