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

"Olle E. Johansson" <oej@edvina.net> Wed, 04 July 2012 15:52 UTC

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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Wed, 04 Jul 2012 17:52:09 +0200
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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4 jul 2012 kl. 17:11 skrev Roman Shpount:

> 
> 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.

Doesn't that mean that the IETF work on SIP over websockets will not lead anywhere?

/O

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