Re: [MMUSIC] Question about Bundle and Legacy Interop (RE: Bundle, TURN and Legacy Interop)

[[e-mail] (Dale R. Worley)]

> From: Paul Kyzivat <[e-mail]>
> 
> IMO there is an (at least implicit) goal that a sufficiently functional 
> sip device be able to interoperate with an RTCWEB-based device without a 
> media gateway.

As Jonathan Rosenberg said
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg06856.html)
> For many application developers, what is interesting about webRTC is
> NOT just browser to browser communications, but connecting users in
> a browser to users elsewhere - on other devices, in other
> networks. After all, the vast majority of web application developers
> do not have the luxury of a massive social graph, or the luxury of
> their users being parked persistently on their website and thus able
> to receive an incoming call. Many websites that have customer
> support or service needs would love to be able to allow their users
> to have a video call with an agent. However, those agents are
> probably sitting on existing agent systems which are not web based,
> and itbTo: 
Subject: 
--text follows this line--
s almost certainly true that they do not today support VP8,
> but rather H.264.

What he didn't say explicitly is that the existing agent systems use
SIP.  So there is a large market to be tapped if WebRTC connects
smoothly to SIP.  And when I say "smoothly", I mean that the web
server/SIP gateway only has to handle "signaling", that it can hand
the SDP across with little or no modification, and doesn't have to
touch the media packets at all.

Of course, WebRTC shouldn't be made less efficient or effective just
to interoperate with SIP, but I don't see that there's any necessity
to do that.  (As long as we remember EKR's dictum that in the context
of negotiating video media, an extra few hundred bytes added to an SDP
description is not significant!)

Dale