Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote recording - RTC-Web client acting as SIPREC session recording client]

"Olle E. Johansson" <oej@edvina.net> Wed, 07 September 2011 20:15 UTC

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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Wed, 7 Sep 2011 22:17:17 +0200
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote recording - RTC-Web client acting as SIPREC session recording client]
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7 sep 2011 kl. 19:43 skrev Harald Alvestrand:

> On 09/07/11 19:29, Ravindran Parthasarathi wrote:
>> 
>> Matthew,
>>  
>> When I asked for SIP, I have meant RFC 3261 only.
> This is 269 pages, and has 26 normative dependencies including S/MIME. It's not a small dependency.
Agree. That RFC is way too focused on ONE application, the phone call.

>> 
>> 4)      There is no need to build two different signaling logic in VoIP servers for each service. Your own example of Bridge line appearance will need two implementation by the single vendor because desktop application or hardphone implementation based on SIP and browser based implementation is depend on HTTP metadata. It is possible to avoided by having one signaling protocol.
> One protocol can be achieved by multiple applications choosing to implement one protocol.
> It doesn't have to be enforced by the prtoocol being embedded in the browser.
Right. 

>>  
>> In RTCWEB does not standardize the signaling protocol interface between browsers, the interop across webservers is next to impossible and it will be restricted to single webserver (company) only. Please let  me know in case I’m missing something here.
> 
> I think the main thing you are missing is that there are many applications that people want to build on top of RTCWEB that are not telephones, and will not fit with telephone-based paradigms.

SIP has already been integrated with years of work of telephone-based paradigms. It's there and we have it. Let's not force that into RTCweb because it will make it way too complex and be an issue when it comes to interoperability between browsers, because most will implement selected parts only (like in SIP today). We don't want that.

I already face that interoperability problem each time I attend SIPit - which btw is open for registration at www.sipit.net for all of you that work with development of SIP apps and devices and need to test interoperability ;-) 

/O