- From: Justin Uberti <juberti@google.com>
- Date: Tue, 23 Jan 2018 22:57:01 -0800
- To: Iñaki Baz Castillo <ibc@aliax.net>
- Cc: Peter Thatcher <pthatcher@google.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>
- Message-ID: <CAOJ7v-3L-tKZ2dQC_w3NJ57OJNy-qEMDaYwDcx3TO9uRfgA8_w@mail.gmail.com>
On Tue, Jan 23, 2018 at 3:37 PM, Iñaki Baz Castillo <ibc@aliax.net> wrote: > On 23 January 2018 at 23:55, Peter Thatcher <pthatcher@google.com> wrote: > > What do we want to accomplish with WebRTC next, and how are we going to > get there? > > This is just a preliminary response since I expect to provide a more > detailed one. > > I don't know which exact additions I'd like to see in WebRTC NG > (although those in your initial email look really nice) but I do know > what I do NOT want to see in WebRTC NG: > > SDP. > Yes, we know. I think that most productive path forward would be to support proposals that you like, rather than continually grinding axes. > > Honestly, I've lost interest in the spec due to the infamous time > spent on SDP related stuff. I just cannot stand with so many email > threads, drafts, proposals and GitHub issues regarding "transceiver", > "direction" attribute in the m-line, the a=msid hack, the ugly remote > track.id value in the SDP, the transceiver.receiver.track > automatically created without having any real remote track at all > (just because `track` is non nullable in RTCRtpReceiver and `receiver` > is non nullable in RTCTransceiver), nor I can stand with so many > discussions about BUNDLE and the issues if a m-line does not provide > ICE candidates and it's not in the BUNDLE a=group, or peerconnections > closing the ICE+DTLS when there are no active send/recv streams just > because SDP was designed that way 20 years ago. > > It's 2018 and people in the rtcweb WG is still discussing (or > re-defining) how codecs are negotiated and re-negotiated in the SDP > O/A protocol. Really? > > Also, I want an API that lets me know what happens *in context*, > meaning that I don't want to rely on contextless "events" fired after > calling an API call such as setRemoteDescription(). I don't want to > receive a blob via my custom signaling, pass such a blob to the WebRTC > API and then wait for events, events that are hard to correlate with > extra information received via sig. And I don't want to do magic to > associate each remote stream/track with the appropriate "user" by > "parsing" the SDP in local or remote side or similar hacks. > > More about this here: https://webrtchacks.com/ > webrtc-sdp-inaki-baz-castillo/ > > > If we really want to make WebRTC useful for developers other than > browser vendors and C++ experts, we cannot transmit a SDP blob as the > "minimum irreducible unit for exchanging multimedia information". > > WebRTC NG must come with a real low level RTC API, and ORTC is a good > place to start (or continue). > > PS: And I want a W3 spec that conforms to a normal W3 spec (rather > than being splitted into a W3 spec plus a IETF spec, JSEP, both > describing how the RTCPeerConnection API works). > > > -- > Iñaki Baz Castillo > <ibc@aliax.net> > >
Received on Wednesday, 24 January 2018 06:57:48 UTC