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

"Asveren, Tolga" <tasveren@sonusnet.com> Tue, 06 September 2011 19:23 UTC

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Date: Tue, 6 Sep 2011 15:24:43 -0400
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Thread-Topic: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]
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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: "Olle E. Johansson" <oej@edvina.net>, "Matthew Kaufman" <matthew.kaufman@skype.net>
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]
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What about semantics (adding just a X-header won't help there) or how much SIP would be left anyhow if all semantical control is exposed through the API?

I think bridged line appearance is a good test to run against different models.

Thanks,
Tolga


-----Original Message-----
From: rtcweb-bounces@ietf.org on behalf of Olle E. Johansson
Sent: Tue 9/6/2011 2:47 PM
To: Matthew Kaufman
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as	SIPREC	session recordingclient]
 

6 sep 2011 kl. 20:40 skrev Matthew Kaufman:

> On 9/6/11 11:36 AM, Ravindran Parthasarathi wrote:
>> Matthew,
>>  
>> Even in case of SIP, there is no need of standardization in case it is within single webserver(skype). SIP supports x-header or proprietary header to extend the way you want it for providing innovative functionality.
> 
> Not if SIP is baked in to the browser it doesn't. If the browser implements a SIP phone in native code, I have no way of adding "x-header" or anything else. I live with the functionality provided by the browser.
That's an implementation detail. One can easily add an API call to add headers on the outbound INVITE.

> 
> If on the other hand, you want SIP implemented in Javascript then sure, if a developer wants to use it, great. If they want to extend it, that's fine too. And if they'd rather use another model, they are free to do that. Just as you would expect from a *platform*.
> 
>> There is no extension barrier from SIP perspective but it ensure that basic call works across web servers. For example, skype browser user will able to talk to gmail real-time user even though all skype functionality are not supported.
> See above.
>>  
>>  In case you have the points why HTTP allows innovation but SIP will not, let us discuss in detail.
> 
> See above. I want you to be free to use SIP, but not *required* to use SIP.
> 
> And there's some security reasons that I'd rather you tunnel it over HTTP rather than opening up your ability to send UDP packets to port 5060.
SIP runs on TCP and TLS ports too.

I am not arguing for either side, just correcting some details...

/O

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