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

[Lishitao <[e-mail]>]

I got some questions about this RTP Retransmission staff, is there a capability negotiation mechanism provided in RTP Retransmission, like negotiate the capability in the first offer/answer transition. (I don’t find that in the documents, or if I miss somewhere)



I also find  in RFC 4588, it mentioned that there are two ways for Retransmission, one is Session-Multiplexing, another is SSRC-Multiplexing, does those two mechanism both need to be supported for implementation?



Regards

shitao



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发件人: [e-mail] [mailto:[e-mail]] 代表 Roman Shpount
发送时间: 2012年7月5日 3:40
收件人: Olle E. Johansson
抄送: [e-mail]
主题: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED


On Wed, Jul 4, 2012 at 11:52 AM, Olle E. Johansson <[e-mail]<mailto:[e-mail]>> 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.
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Roman Shpount