[RTW] [dispatch] RTC-Web I-D about interworking between RTC-Web and SIP-RTP

[Adam Roach]

On 2/9/11 13:52, Feb 9, Markus.Isomaki at nokia.com wrote:
> The RTP/media stack on the other hand is definitely in the scope of 
> the IETF effort. I think we should standardize the RTP use in the 
> browsers and that would be one step towards interop with SIP phones. 
> The critical thing seems to be the STUN connectivity check or media 
> authorization part. If we mandate browsers to get that exchange done 
> before they are allowed to generate any RTP packets on behalf of the 
> application, this will ruin the possibility of interop with 99% of 
> existing SIP clients (without some kind of an SBC). DTMF transport 
> capability may also be relevant interop requirement.

+1.

Also, it would be incorrect to characterize RTP as being solely for the 
benefit of SIP. Several other protocol -- both published and proprietary 
-- make use of RTP; Jingle comes to mind, as do the proprietary 
protocols used by various IP PBX vendors.

If we decide to go down an "RTP but with special sauce" or "not quite 
RTP" path, we lose a lot of potential functionality.

/a
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