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

Hadriel Kaplan <HKaplan@acmepacket.com> Mon, 14 November 2011 12:38 UTC

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Neil Stratford <neils@belltower.co.uk>
Thread-Topic: [rtcweb] Traffic should be encrypted. (Re: Let's define the purpose of WebRTC)
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Date: Mon, 14 Nov 2011 12:38:32 +0000
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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 Nov 14, 2011, at 6:48 AM, Neil Stratford wrote:

> In my example the terminating WebRTC media server *is* the PSTN gateway. There is no RTP beyond the gateway, just ISDN etc. In this case to get the most reliable DTMF transport from the client to the PSTN I'd have to roll my own DTMF transport over a DataStream, carry if over the signalling channel, or accept the lossy RTP DTMF channel. In many cases this DTMF RTP will be the only RTP sent in that direction - I'm often asked for the ability to send DTMF without requesting microphone access permission for information only IVR use cases.

If you create your own media server, you can do whatever you want - you don't need the IETF for anything.  I mean you can open a SCTP/DTLS/UDP data-channel and send whatever you want over it; or you can open a websocket to it and send whatever you want over it, etc.  Why would you even use "DTMF" for that - i.e., why would you constrain yourself to only a few digits/characters?

The only reason to have "DTMF" as true "DTMF" a la rfc4733 is to be able to use media-servers created from other vendors/third-parties, possibly not even managed/owned by the same domain as WebRTC is running in.  But for that to work, it needs to be based on a standard.  The only two IETF standards for that today are KPML and RFC4733.  And if we want it to be based on a standard most media-servers actually implement, that would be RFC4733.

-hadriel