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

Roman Shpount <roman@telurix.com> Wed, 04 July 2012 15:11 UTC

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Date: Wed, 04 Jul 2012 11:11:29 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On Tue, Jul 3, 2012 at 3:30 AM, Olle E. Johansson <oej@edvina.net> wrote:

>
> 2 jul 2012 kl. 22:18 skrev Justin Uberti:
>
> > As has been pointed out in this thread before, this discussion is not
> about mandating the USE of retransmission in realtime scenarios.  It is
> simply trying to decide whether retransmission should be required to be
> present in the 'toolbox' of tools that WebRTC apps can expect to use,
> primarily where the app developer and runtime developer are separate (e.g.
> in a browser).
>
> I fully understand this. What worries me is that my experience from SIPits
> and working with SIP for many years is that there is a large gap between
> IETF documents and what the implementations out there support. I would like
> some base level of interoperability in the RTP layer between webrtc and the
> SIP installed base. If the RTP toolset for Webrtc is far away from any
> standard SIP phone, you will be forcing everyone to use application layer
> gateways. I think that would be a bad scenario.
>
> I still think it has to be optional to implement.
>
>
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.
_____________
Roman Shpount