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

Matthew Kaufman <matthew.kaufman@skype.net> Tue, 06 September 2011 21:23 UTC

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Date: Tue, 06 Sep 2011 14:25:07 -0700
From: Matthew Kaufman <matthew.kaufman@skype.net>
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Cc: rtcweb@ietf.org
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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On 9/6/11 6:13 AM, Igor Faynberg wrote:
> I find this reasoning hard to understand.  First, the SIP standard has 
> been around for a decade. What new standardization is needed?

The issue is whenever you wish to innovate and add functionality.

See the long, long list of RFCs that extend SIP. For some examples:
RFC3515 - Refer method
RFC3842 - Message waiting
RFC3891 - Replaces header
RFC3911 - Join header
RFC3960 - Early media
RFC4411 - Reason header for preemption

Never mind all the things that aren't even finished...

draft-ietf-bliss-shared-appearances
draft-ietf-bliss-call-park-extension

true bridged line appearances

etc

etc

If all you want is a basic point-to-point phone, there's a short list of 
RFCs you implement and you're done. But that isn't a platform for 
innovation... it is just a phone.

Matthew Kaufman