Re: [rtcweb] Traffic should be encrypted. (Re: Let's define the purpose of WebRTC)

Roman Shpount <roman@telurix.com> Mon, 14 November 2011 14:49 UTC

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Date: Mon, 14 Nov 2011 09:49:02 -0500
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From: Roman Shpount <roman@telurix.com>
To: Neil Stratford <neils@belltower.co.uk>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Traffic should be encrypted. (Re: Let's define the purpose of WebRTC)
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On Mon, Nov 14, 2011 at 6:48 AM, Neil Stratford <neils@belltower.co.uk>wrote:

> I'd prefer we didn't have a DTMF API and instead leave it up to the
> developer to send over the signalling channel (the usual sync arguments
> don't apply here if you are sending from an API), but if we do have an API
> it makes sense to use the most reliable transport available to carry it.
>

Even if WebRTC is not going to support RFC 4733 DTMF (I am not saying it
should not. I would strongly prefer it did.), we still need to deal with
timing sync between commands and media content. Imagine an application
which asks to press "Done" after recording is complete. If there is no
timing sync, "Done" command can arrive plus/minus half a second from the
end of an actual recording. This makes a lot of applications, like record a
greetings almost unusable. I think that even if WebRTC will support DTMF,
it would be useful to have some type of API which will allow to time sync
arbitrary signaling messages with media stream. One option would be to give
JavaScript access to media stream time stamp. Another option (less
preferred from my point of view) is to have a separate RTP payload that can
be used to transport arbitrary application messages.
_____________
Roman Shpount