Re: [rtcweb] Let's define the purpose of WebRTC

Hadriel Kaplan <HKaplan@acmepacket.com> Mon, 07 November 2011 03:08 UTC

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] Let's define the purpose of WebRTC
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Date: Mon, 07 Nov 2011 03:08:23 +0000
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Subject: Re: [rtcweb] Let's define the purpose of WebRTC
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Hey Tim,
comments inline...

On Nov 6, 2011, at 9:01 AM, Tim Panton wrote:

> Almost all of the discussions here in the last few weeks are about interop with existing SIP 
> deployments. ( forking,glare, SDP, RTP muxing)

RTP muxing is actually not an existing SIP thing - it's new.  SDP obviously is an existing mechanism and used in SIP, but if we're going to have a high-layer abstracted "API" anyway regardless of SIP interop, then at least SDP is an already existing/defined protocol.  Glare and forking could both be handled with purely application coding (ie, javascript), so yeah those are more of a conciliation for interop.


> The bulk of the use-cases on which that effort is
> based are also around interop with non-browser devices and channels, but the charter
> never mentions legacy interop. It only talks about browser-to-browser and replacing the
> myriad plugins. It also talks about being a platform for innovation, which is now a
> non-goal for the group.

Actually, in the charter itself is a list of work the WG will perform, and item 9 says this:
      The group will consider options for interworking with legacy VoIP
      equipment.

I think the reason a lot of the email discussions have been focused on that is that it's the most contentious area of issues to solve or make decisions about.  Obviously some folks also see a reasonable use-case for interworking with legacy VoIP, but so long as non-inteworking/WebRTC-only use-cases aren't impaired by supporting interworking, what's the harm?

-hadriel