16:06:01 RRSAgent has joined #webrtc 16:06:01 logging to https://www.w3.org/2022/12/07-webrtc-irc 16:06:07 scribe: hta 16:06:14 regrets+ Dom 16:08:39 ... survey shows strong support for use of async funtions 16:09:24 ... most devs munge SDP 16:10:40 ... 1/3 of respondents don't use datachannels 16:10:57 ... many want to do datachannels in workers 16:12:02 ... survey was anonymous (collecting ID would need GDPR stuff) 16:13:32 ... API regarded as complex or very difficult by most 16:13:47 karstenboehm has left #webrtc 16:15:08 ... survey was focused only on JS APIs, not native 16:17:04 bernard, youenn: SDP munging was the biggest surprise / learning 16:17:52 Tim: we should look to other APIs to see if there are patterns we could adopt 16:20:15 youenn: devtools? WebInspector? 16:21:12 bernard: want to know if there are a few things in SDP munge or many 16:21:28 jan-ivar: not all APIs that avoid SDP munging are implemented in all browser 16:23:39 fippo: 10% of SDP munging is to achieve stereo in Chrome. we don't know who. 16:25:15 fippo: will produce a list of things people can do / are doing with SDP munging. 16:26:32 Tim: should we adopt methods / tricks / constructs from other APIs? 16:27:46 ... Maybe, but not urgently. 16:29:25 Bernard: Presenting scenarios for CfC 16:34:19 hta: 3.2.1 requirements should be in terms of desired results, not in terms of how to achieve them. 16:35:45 youenn: do we want ability to inject JS in the media pipeline, or is latency too critical? 16:37:13 tim: need API to match up server stuff with client stuff 16:39:42 Bernard presents 3.2.2 16:42:04 youenn: points out that N39 kind of presupposes that we want to use RTP 16:44:27 youenn: not clear whether this is frame level or packet level 16:44:59 hta: that choice may be part of the solution space rather than the requirement space 16:45:30 jan-ivar: could also consider the case of within-browser relay (no js). 16:47:27 pthatcher: running over datachannel gives worry with congestion control. 16:47:49 bernard: people have done over datachannel, with custom code sender-side. 16:48:53 Bernard presents 3.5 Virtual reality gaming 16:49:35 youenn: pre-sending video frames & metadata may be requirements. Reach out to XR? 16:50:58 Harald's presentation on issue 106 16:51:25 hta: original use case was e2ee 16:52:29 ... slide 28 16:52:45 ... this is Bernard's summary 16:53:39 Tim: it's not necessary about slowing or speeding things 16:54:01 hta: [slide 29] 16:55:01 ... request from people feeding H264 to a browser using something else than webRTC, that they want to send over RTP 16:55:30 ... we can do that without using the encoder 16:56:07 bernard: I heard that use case for traffic cameras 16:57:30 jan-ivar: it's hard to gauge if it can't be done with the existing APIs 16:57:57 youenn: I'm still fuzzy about this UC 16:58:21 ... it seems that one node in the network is producing H264 and it gets sent to a browser 16:58:45 ... I'm not sure why there should be a browser in the middle between the sender and the RTP 16:59:08 bernard: it's coming over HTTP most of the time 16:59:35 youenn: it's not the typical browser use 16:59:50 ... the context would help understanding the UC 17:00:11 hta: the summption is that the camera is not supported by GUM 17:00:20 s/summ/asum/ 17:00:31 s/GUM/gUM 17:00:49 jan-ivar: is the UC supporting traffic cameras in the browser? 17:01:17 ... how do you feed to the browser? 17:02:29 Tim: it would be nice to have something to prevent re-encoding 17:02:56 ... If you have an in-car camera with limited BW to a monitoring station 17:03:25 ... and that station is a browser, but you want to share the feed to someone else over webRTC 17:03:48 ... rendering + capturing/sending is overkill 17:04:35 jan-ivar: if low latency is the goal (thus sending to RTP), @@@ 17:05:21 hta: if you take away h264, there are other solutions 17:05:35 hta: [slide 30] 17:06:38 ... this UC is pre recorded media sent to RTP 17:08:12 youenn: I don't understand why the desired tranceiver would be RTP 17:08:33 ... for that UC to be meaningful you need to know why RTP is better 17:08:55 hta: the scenario could be that you already have an RTP cnx 17:09:02 ... e.g. live video 17:09:24 ... then you want to send pre-recorded video, without the receiving side having to do anything 17:09:43 jan-ivar: you could do this using the video track generator 17:09:58 ... you could decode the pre-recorded 17:10:14 ... so this (proposal) is an optimization 17:11:06 hta: we can switch to key frames. it's UC dependent whether we want the browser to do it 17:11:30 bernard: there might also be battery-life optimizaiton 17:11:56 youenn: it would be good to comment in the issue if we're targeting optimization 17:12:03 hta: please comment 17:12:24 Peter: [slide 31] 17:14:17 youenn: should we have an API that controls at the packet-level or an API at the encoder side 17:14:41 ... I think we be trying to do both in the same API and it's too difficult 17:15:02 ... are there 2 different sets of UCs, e.g. low-latency and other optimizations 17:15:12 Peter: maybe true 17:15:51 youenn: webRTC NV could have those UCs and requirements, do we need packet data API, or frame 17:16:18 Peter: @@@@ audio codecs 17:16:43 youenn: look at what we expose 17:17:18 Topic: Encoded media manipulation 17:18:08 hta: [slide 35] 17:19:17 hta: [slide 36] 17:20:29 hta: [slide 37] 17:22:08 hta: [slide 38] 17:23:13 hta: I'd like to have a Call for Consensus on the low latency streaming UC 17:24:18 Tim: you need to have a rough flavor of an API 17:25:01 ... who are targeted audiences? does that fit naturally with codecs? 17:25:15 ... at least describe that is useful 17:25:41 hta: the obvious APIs to fit next to are webRTC APIs and webCodec APIs 17:26:01 jan-ivar: we don't have to agree to incremental APIs 17:26:49 hta: if I have something that is part of solution, we don't need the entire solution before pushing it 17:27:22 jan-ivar: it sounds OK but I don't think we need to commit to this approach 17:27:34 ... in advance 17:27:59 youenn: i'd echo what Tim said, we need to understand the big picture of what we're trying to solve 17:28:26 ... and I agree it's good to look at webCodec APis 17:29:03 ... webRTC encoded transform is fine 17:29:27 ... we should design the best API, not the API that needs the less work for the User Agent 17:29:44 Topic: issue 143/PR 165 17:30:01 Fippo: the proposal is to merge PR 165 17:30:19 ... [slides 43-44 showing the PR] 17:30:32 hta: I agree 17:30:57 youenn: we discussed rids but not the return value 17:31:24 ... the return value is useful to validate the input 17:32:07 ... the promise makes sense for the transformer 17:32:25 ... I don't have a strong opinion for the sender 17:32:52 fippo: you can wait and read the key frame 17:33:08 youenn: with the promise you can reject asynchronously 17:33:14 ... it helps a bit 17:33:33 jan-ivar: the PR removes a lot of the algo. text 17:34:07 youenn: the text detailed all the cases 17:34:24 ... if you remove the promise resolution you can simplify a lot 17:34:40 jan-ivar: we may not merge PRs during meetings 17:34:58 youenn: we did not yet agree about the return value 17:35:11 ... so we need to decide 17:35:34 bernard: objection to no return value? 17:35:57 youenn: original proposal was a promise 17:36:17 ... safari implements the promise undefined 17:36:36 hta: no objection to using that 17:36:45 Topic: issue 167 17:36:55 fippo: [slide 45] 17:37:30 ... proposal is to add timing metadata 17:37:49 ... currently only things derived from rtp header 17:38:43 bernard: this is an issue, we try to give a direction 17:39:06 youenn: I have not looked beforehand, no input yet 17:39:18 fippo: pls leave feedback on the issue 17:39:30 Topic: issue 168 17:39:42 fippo: [slide 46] 17:40:27 ... I'd like to add rtp header metadata 17:40:54 hta: are they read only? 17:41:45 fippo: [slide 47] 17:42:23 ... [slide 48] 17:42:38 ... the PR is only for audio at the moment 17:42:47 ... can we merge this PR 154? 17:44:14 Tim: I'll lokk at the PR 17:44:21 s/lokk/look 17:44:51 jan-ivar: no consensus yet 17:45:43 fippo: [slide 49] 17:46:05 ... we should not use SSRC 17:46:24 ... so proposal to add mid and rid to metadata 17:49:25 Action: Fippo to provide a PR 17:49:46 fippo: [slide 50] adding timestamp 17:51:36 OK. 17:51:52 scribe: hta 17:52:12 fippo: [slide 51] 17:52:38 ... mimeType metadata 17:53:03 ... add fmtp line? 17:53:20 bernard: webcodecs has them both in one object 17:54:02 youenn: like the webcodecs way better 17:55:08 fippo: complex to get access to the fmtp line in encoded-transform 17:58:35 fippo: discussing this further. 17:59:27 bernard: [slide 53] 18:00:43 ... describes SVC metadata; webcodecs is "everything from DD", encoded-media has type mismatches, missing info 18:02:20 ... suggests using a subdictionary for SVC information in metadata 18:04:27 what's the command for generating miutes again? 18:04:40 Present: Bernard, Harald, Jan-Ivar, Patrick Rockhill, Peter Thatcher, Philipp Hancke, Tim 18:04:42 Panton, Youenn, Carine, Brian Baldino, Florent Castelli, Palak 18:05:03 RRSagent, make log public 18:05:28 present+ Youenn, Carine, Brian Baldino, Florent Castelli, Palak 18:05:35 RRSAgent, make minutes 18:05:35 I have made the request to generate https://www.w3.org/2022/12/07-webrtc-minutes.html caribou 18:07:04 i/Harald's presentation on/scribenick: caribou 18:07:07 RRSAgent, make minutes 18:07:07 I have made the request to generate https://www.w3.org/2022/12/07-webrtc-minutes.html caribou 18:07:48 s/Panton, Youenn, Carine, Brian Baldino, Florent Castelli, Palak// 18:07:56 s/what's the command for generating miutes again?// 18:08:05 RRSAgent, make minutes 18:08:05 I have made the request to generate https://www.w3.org/2022/12/07-webrtc-minutes.html caribou 18:08:47 RRSAgent, stop