15:04:02 RRSAgent has joined #webrtc 15:04:07 logging to https://www.w3.org/2026/03/24-webrtc-irc 15:04:07 Zakim has joined #webrtc 15:04:13 Chairs: jan-ivar, guido; youenn 15:04:17 s/;/,/ 15:04:29 Recording is starting 15:04:29 Recording: https://www.youtube.com/watch?v=TilVQOs8WPY 15:04:34 Meeting: WebRTC March 2026 meeting 15:04:44 Agenda: https://www.w3.org/2011/04/webrtc/wiki/March_24_2026 15:04:59 Slideset: https://docs.google.com/presentation/d/1ECkxmWxXJFK_mv14UHcLiDmO6KczaHQhTVlE69lAqEs/ 15:05:59 Present+ BartoszHabrajski, Guido, Harald, Jan-Ivar, KacperWasniowski, PeterT, SunShin, TimP, Youenn, Dom 15:06:11 Present+ ThomasNGuyen 15:06:40 present+ 15:07:54 Topic: -> https://github.com/w3c/webrtc-pc/issues/ WebRTC-PC Interop Issues 15:07:54 [slide 10] 15:08:06 Subtopic: Issue #3088 [[IceRole]] initialized too late 15:08:09 [slide 11] 15:08:56 [slide 12] 15:09:58 [slide 13] 15:10:43 [no reaction] 15:11:04 s/[no reaction]/Youenn: any observable change from this? 15:11:35 Jan-Ivar: I don't think so, but still to be confirmed through updated WPTs 15:11:48 RESOLVED: Proceed with PR to fix this 15:12:05 Guido: the PR may provide opportunity for more detailed reaction 15:12:07 Subtopic: Issue #3090 · Should garbage collecting RTCPeerConnection be observable? 15:12:07 [slide 14] 15:17:42 Youenn: when eventhandlers are registered, the PC isn't GC-able 15:18:25 ... pc.onicecandidate is registered in the test under discussion 15:18:51 Harald: I'm remembering updating tests to add pc.close() to ensure GC 15:19:15 ... we've been pushing that any way of reaching an open PC should make it not GC-able 15:19:46 Jan-Ivar: GC is obversable from the remote end 15:22:32 ... current Firefox behavior is to GC more aggresively than other browsers, which avoids issues when devs don't call pc.close() 15:23:59 Youenn: in the test case we're discussing, this is because FF detects no ice candidate can reach pc2? otherwise, it feels like a bug in FF 15:25:53 Youenn: not sure about changing the spec; the test probably needs more review 15:26:04 Dom: how much real-word compat issue is this creating? 15:26:09 Jan-Ivar: not aware of any specifically 15:26:20 Dom: then maybe we don't need to fix it 15:26:36 Harald: fixing the test feels like always worth it 15:26:44 Subtopic: Issue #3092 · RTCError should be exposed on Workers 15:26:44 [slide 15] 15:27:48 RESOLVED: adopt proposal 15:27:57 Subtopic: Issue #3095 · https://w3c.github.io/webrtc-pc/#dfn-final-steps-to-create-an-offer is unclear 15:27:57 [slide 16] 15:31:46 [slide 17] 15:33:57 Jan-Ivar: another option is to clarify the intent of the spec 15:34:01 Youenn: that's option A 15:34:48 Jan-Ivar: this predates promises, when we had concerns about the negotiationneeded event 15:35:40 ... this may no longer be as much an issue if you await promises, but there are still hard-to-pin-down tricky timing issues that this was meant to prevent and are likely still relevant 15:36:56 ... FF optimizes this cleverly, which could serve as a basis for a more specific algorithm 15:37:40 ... the main goal is to ensure that as you're about to send the results of createOffer, you ensure it's still aligned with JS state 15:38:07 Youenn: given the example is about system resoures, that's not so clear to me 15:39:15 Harald: the relevant test creates an offer without any tracks, then add a track; this is not obvious that this is what the spec is asking 15:40:13 ... I'd be in favor of some version of Option A if we can find a clean way to spec it 15:41:02 Youenn: if we go with option A, if negotiation might be needed and createOffer is in process, you would need to check 15:43:29 ... the spec already asks to generate the answer on main thread 15:46:03 ... we should either clarify the spec with normative language, or non-normative language, to ensure alignment between what's the page is asking and the offer 15:46:19 RESOLVED: Proceed with option A, ideally with normative language, or non-normative if not possible 15:47:12 Guido: is this causing issues in the wild? if not, option C is good 15:47:22 Youenn: not aware of real web compat issues 15:47:42 Jan-Ivar: can't rule it out since they would be really hard to track down 15:49:29 s/RESOLVED/PROPOSED/ 15:49:53 Harald: let's see a concrete proposal for option A before resolving this 15:53:38 TimP: if the change you've put in JS is not in the offer, won't there be a negotiationneeded event to fix things? 15:53:55 Jan-Ivar: but using negotiationneeded isn't required 15:55:46 TimP: if this was designed before we added better APIs (negotiationneeded and transceiver), is this complexity still needed? 15:56:12 Jan-Ivar: there are data races that the spec should protect from, even if they're corner cases 15:57:20 youenn: it's not an alternative between race and no-race - it's mostly about optimizing API shape 15:57:30 Jan-Ivar: I'll provide more background on the historical intent 15:57:43 Subtopic: Issue #3096 · Vague error for setting a description, inconsistent wpt 15:57:43 [slide 18] 15:59:16 [slide 19] 16:02:00 [slide 20] 16:02:37 Guido: does this mean Chrome and Safari would fail the test? 16:02:40 Jan-Ivar: yes 16:03:21 Harald: I agree with the principle 16:04:03 Guido: I'm OK with fixing the test, but wouldn't want to keep it in Interop 26 16:04:38 Youenn: I'm OK with the fix; no strong feeling about interop 26, doesn't look like it should be a big change to libwebrtc 16:05:40 RESOLVED: Proceed with fixing test case; integration in interop 26 still TBD 16:05:58 Subtopic: Issue #3097 · Should RTCDataChannel.send be allowed to throw synchronously for not enough buffer space 16:05:59 [slide 21] 16:08:23 [slide 22] 16:10:56 Jan-Ivar: FF lets you send an unlimited amout since we didn't see a reason to limit it; the proposed change works for us, although UA-dependent behavior isn't great 16:11:28 Youenn: this may be something worth bringing in a separate issue, but not part of interop 26 16:11:36 RESOLVED: Proceed with proposal of spec clarification 16:11:55 Topic: Capabilities Element 16:11:55 [slide 25] 16:12:42 s/Topic: Capabilities Element// 16:12:47 s/[slide 25]// 16:12:56 Topic: Capabilities Element 16:12:56 [slide 25] 16:13:27 Daniel: we've been discussing the element, now I have a draft spec 16:13:58 ... interested in guidance in how to proceed with it moving forward 16:14:14 ... we have a chrome legacy attribute on which I'm also seeking guidance 16:15:11 Jan-Ivar: going piecemeal would be OK 16:15:46 Youenn: +1 with going piecemeal after initial explainer - you may need to update mediacapture-main with hooks 16:16:07 Daniel: should I model this after the other explainer in mediacapture-extensions? 16:16:50 Guido: start from the TAG model 16:16:51 -> https://w3ctag.github.io/explainer-explainer/ Explainer Explainer 16:17:20 Youenn: so start with an explainer, then a series of PR on media capture extensions (possibly completed with additions to -main as needed) 16:17:34 ... do you have a sense of a timeline? 16:17:55 Daniel: hope to get back to this next week; we already have a Blink-process explainer which I hope can serve as a good basis 16:18:28 Topic: MSTP audio / AudioTrackGenerator 16:18:28 [slide 28] 16:20:16 [slide 29] 16:21:03 [slide 30] 16:22:45 Jan-Ivar: the data has been longer to produce than initially planned given Paul's availability 16:23:15 TimP: I have some sympathy on the overlap argument, but the use cases are quite different 16:24:07 ... MSTP is mostly useful when you're processing both audio and video together - having different APIs for things that are tightly coupled in the way developers approach it is not very satisfactory 16:25:18 ... e.g. do lips-rerendering in video based on the audio 16:25:47 Jan-Ivar: our finding is that audio and video are different enough that you're likely to end up wanting to treat them differently 16:26:00 TimP: I still feel like there is a useful niche in approaching them together 16:27:53 Guido: beyond measurements (on which I'm looking forward to refutal) - there is a non-niche use case requested by multiple use cases we presented at a previous meeting where audio glitches feel unavoidable when processing doesn't hit audio deadlines 16:29:40 ... this particular use case can't be implemented without glitches in FF and Safari given lack of MSTP audio 16:32:00 Jan-Ivar: might be worth highlighting it on the github issue 16:33:07 Youenn: TimP made a good point about the scenario where having main thread + audio worklet + worker - this would be what a test should demonstrate 16:33:48 ... it would be good to ensure the requirements from Chrome's team are well understood 16:37:13 Guido: we have sample code that illustrated our use case 16:38:02 Youenn: so two actions are needed: a better MSTP shim, and an illustration of how to avoid glitches when audio processing can't meet the audio clock based on the sample Guido shared previously 16:39:13 Harald: I would encourage a dedicated meeting between Guido and Paul to advance on this outside of a WG meeting 16:40:06 -> https://www.w3.org/2025/12/09-webrtc-minutes.html#97b7 Previous discussion on the topic 16:41:15 Jan-Ivar: to clarify, the goal of the shim is not to demonstrate no perf impact - the proper comparison should be with a audio worklet architecture 16:41:48 TimP: if there is a such meeting, hope the group is invited (even if not required) 16:43:41 @@@: if there are differences of jitter between input and rendering, there will be glitches if there is no buffering, given WebAudio and MSTP have different models (push vs pull) 16:45:25 Youenn: what timeline for these actions? given potential dependency on implementation work 16:45:37 Jan-Ivar: I need to check with Paul 16:45:54 RRSAgent, draft minutes 16:45:55 I have made the request to generate https://www.w3.org/2026/03/24-webrtc-minutes.html dom 16:51:46 RRSAgent, make log public 18:06:31 cabanier has joined #webrtc 18:21:51 Zakim has left #webrtc