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Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED

Roman Shpount <roman@telurix.com> Wed, 04 July 2012 19:39 UTC

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Date: Wed, 04 Jul 2012 15:39:49 -0400
Message-ID: <CAD5OKxszD1ia=8SwH0fSFDatfQxAMQ500y7ZYBhXqGgNSj28nA@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: "Olle E. Johansson" <oej@edvina.net>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On Wed, Jul 4, 2012 at 11:52 AM, Olle E. Johansson <oej@edvina.net> wrote:

>
> 4 jul 2012 kl. 17:11 skrev Roman Shpount:
>
> >
> > I think this train left the station long time ago. WebRTC is not going
> to interoperate with anything of significance currently deployed in SIP
> world. Between DTLS-SRTP, ICE, AVPF and now RTP retransmissions you will
> need an application layer gateway. I guess the overall hope is that SIP
> implementations will catch up implementing features required to work with
> WebRTC in the future and gateways would be acceptable for current
> implementations.
>
> Doesn't that mean that the IETF work on SIP over websockets will not lead
> anywhere?
>
>
Unless we have a SIP end point that supports DTLS-SRTP  and ICE,
understands AVPF profile, and knows how to negotiate RTP retransmissions
and other RTP extensions, then I would say SIP over websockets will not
help in communications between SIP and WebRTC. There are a few SIP
endpoints that claim to support some of those specifications, but anything
massively deployed in real life, like IP phones, PBXs, SBCs, PSTN gateways,
and soft-phones,  almost universally do not support or use any of those
standards.

SIP over websockets might be useful on its own though, for instance as a
replacement for SIP-outbound, which fails to address even basic operational
requirements for SIP clients behind NAT. Hopefully though, WebRTC will
replace SIP on the client, and will remove the need for SIP behind NAT
support altogether, making SIP over websockets completely irrelevant.
_____________
Roman Shpount

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