Re: it's time to expel SDP from our lives
Iñaki Baz Castillo <[email protected]>
| Newsgroups | gmane.ietf.mmusic |
|---|---|
| Message-ID | <CALiegf=4ANANnM_14Z4QwnS2LiUxmDniyicL56X635ERq5WpkA@mail.gmail.com> |
2017-06-13 22:33 GMT+02:00 Paul Kyzivat <[email protected]>: > Everybody who has worked with SDP realizes that it is ill-suited to the > tasks we use it for. Quite a long time ago there was an effort called SDPng > that tried to define a new syntax. I wasn't involved in that, but it seemed > like it was defining something reasonable (XML based IIRC). But it failed > because (IIUC) nobody could see a viable migration plan to introduce it. > > I think you are suggesting to do something new but limit its scope to > WebRTC. Such a narrow scope would ease the migration effort. There would > still be one but perhaps people would be willing to suffer through it. > > BUT, WebRTC doesn't stand alone. Interworking with SIP is still important, > and probably will be for a long time. Also, your description of what is > needed is simplistic. What features (functionality, not syntax) of SDP O/A > are not needed with WebRTC? AFAICT all the stuff that describes codecs is > needed, and much of the rest. There is a lot of that, spread over many RFCs. > All of that would have to be re-done. > > Doing something new here may well be the right thing to do. But the first > step is to carefully define the scope of applicability, and what > interworking requirements there are with things outside that scope. Then > perhaps the size of the effort can be estimated and the feasibility doing so > assessed. Hi Paul, I'm not asking for a SDP 2.0 in JSON/XML or any other format. In your text above, you state that "all the stuff that describes codecs is needed, and much of the rest", and I don't agree with that: ORTC has proven to provide the *same* functionality as SDP, but without imposing the exchange of a full SDP blob/string. Yes, codec parameters must be exchanged between peers, but that does not mean that a SDP (or similar format) is required. Instead, in ORTC, it's up to the application/developer to decide how to signal parameters between the endpoints of a multimedia session. In ORTC an endpoint can signal ICE parameters, DTLS parameters, RTP parameters, and other parameters in an independent way (so, if for example a new video stream is added, there is no need to signal all the ICE/DTLS stuff to the remote party again, but just the RTP parameters related to the new stream). This design prevents issues such as the "SDP DTLS role conflict" described in the previous email thread. You say "There is a lot of that, spread over many RFCs". And that's true, but ORTC specification [*] collects many of those parameters and re-uses them within their own Dictionaries about ICE, DTLS, RTP, RTCP, etc. It's true that there are tons of RFC's that define multimedia parameters assuming SDP syntax, but IMHO the SDP-ish is just a minor component in those specifications, nothing too hard to circumvent IHMO (in fact, ORTC did it). > I think you are suggesting to do something new but limit its scope to WebRTC Not exactly. Continue reading, please. > WebRTC doesn't stand alone. Interworking with SIP is still important Sure, and I understand that SIP indeed requires a wire-signaling format/syntax because SIP requires interoperability between different devices in both, the signaling and media, planes. But IMHO that does not justify the fact that all the multimedia related specifications are tiled to just SDP. SDP should not be the minimum irreducible unit for exchanging multimedia information, and ORTC is a good example of the opposite. What I would expect nowadays from the MMUSIC WG is to define multimedia parameters (including different *layers" such as transport, security/encryption, RTP, RTCP, etc) rather than a single format(SDP) to cover them all. Having that, WebRTC could perfectly drop SDP and define some proper Dictionaries (as ORTC) pointing to those parameters defined by MMUSIC so WebRTC developers are done and free to exchange data as they wish. And when it comes to SIP, of course the SIP protocol needs a "multimedia format". A separate WG or a different task within MMUSIC could be responsible of producing an standardized "multimedia format" for SIP with those parameters, let it be SDP or SDP 2.0. Anyhow, if so many things within the SDP are so problematic for WebRTC, they will also be problematic for SIP (once SIP devices implement so many advanced features such as those in WebRTC devices), so still I consider that sending a full SDP to just renegotiate a new codec (or remove a specific video stream) will produce real problems as we suffer in WebRTC. But again, this is a separate concern that should just affect to SIP devices (because SIP chose to go with SDP), and not to WebRTC. Thanks a lot for your response, regards. [*] http://draft.ortc.org/ -- Iñaki Baz Castillo <[email protected]> _______________________________________________ mmusic mailing list [email protected] https://www.ietf.org/mailman/listinfo/mmusic