Friday, March 12, 2021< ^ >
martin.thomson has set the subject to: RTCWEB WG
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[11:58:31] <Cullen Jennings_web_828> why do the slides say make sure your video is off ?
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[12:00:25] <Cullen Jennings_web_828> thanks
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[12:01:58] <Eric Rescorla_web_500> Oh fine
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[12:02:14] <Eric Rescorla_web_500> Let me see if I can pull up CODIMD
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[12:02:31] <Sean Turner_web_362> v
[12:02:33] <Sean Turner_web_362>
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[12:03:12] <Eric Rescorla_web_500> annoyingly it just has a spinner when it's in-frame
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[12:04:27] <Eric Rescorla_web_500> meetecho: yeah, the section you type in just has a spinner
[12:04:49] <Meetecho> Eric: do you mean codimd?
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[12:05:09] <Eric Rescorla_web_500> yes.
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[12:05:15] <Eric Rescorla_web_500> When I opened it in a new window it worked
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[12:05:52] <Meetecho> It may have to do with cross-site cookies: it's known not to work if you're running meetecho in a private/incognito window for instance
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[12:07:07] <> Hmm.... Well I do have Firefox tracking protection on, but (1) the UI loads and (2) i don't see any errors in the browser console
[12:07:36] <> And TP doesn't block non-trackers, at least theoretically
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[12:10:44] <> So I do think we should consider this a potential bug in the site
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[12:11:08] <> though perhaps not something that Meetecho needs to concern itself with
[12:11:56] <Meetecho> Ekr: we'll try to debug your specific scenario when the event is over, to see if we can get codimd to work even in those scenarios. Thanks for the heads up!
[12:12:02] <> thx
[12:12:25] <> Happy to do some debugging. Like I said, I don't see anything in the browser console with errors, but maybe i missed it
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[12:20:43] <Lorenzo Miniero_web_865> Bernard: it can lead to issues actually, e.g., we had some similar issues interacting with gstreamer's webrtcbin because of that. You don't always have a JavaScript app where you can tweak things.
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[12:31:26] <Jonathan Lennox_web_478> Does Firefox also exhibit the same behavior as libwebrtc?  (Not sure whether this behavior comes from the part of libwebrtc Firefox uses or not.)
[12:31:48] <> No, itdoes not
[12:32:08] <Jonathan Lennox_web_478> Which behavior does it implement?
[12:32:13] <> The JSEP one, I believe
[12:34:19] <> @Jonathan Lennox: the answer to your more general question is that the only pieces we share with libwebrtc are about the media proper
[12:36:44] <Cullen Jennings_web_828> Thanks tim
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[12:41:18] <Lorenzo Miniero_web_325> Roman: you need to click the mic icon to talk
[12:46:48] <hta> Datachannel apps are probably 100% written to webrtc, and likely all do bundle.
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[12:54:01] <hta> I like must-bundle :-)
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[12:58:41] <> must-bundle-safe
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[13:06:53] <> hey chairs, are we done with this topic?
[13:07:26] <Ted Hardie_web_860> @ekr Roman is the last speaker, as far as the queue goes.
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[13:11:55] <Jonathan Lennox_web_478> Would we want this to be a full bundle-bis, or a patch document?
[13:15:06] <> LIVING DOCUMENTS!
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[13:16:30] <Timothy Panton_web_296> At the risk of diving into the weeds, can we avoid using max-bundle-safe and use something less negative ?must-bundle  perhaps?
[13:16:56] <Christer Holmberg_web_151> I suggested must-bundle in a private discussion
[13:17:00] <Cullen Jennings_web_828> Huge thanks to Christer for doing the work on this.
[13:17:05] <Sean Turner_web_362> The important point about this is that there is no opposition ... great!
[13:17:18] <Suhas Nandakumar_web_854> +1
[13:17:19] <Sean Turner_web_362> @Christer yes thanks
[13:17:20] <Cullen Jennings_web_828> +1 on must-bundle as label
[13:17:34] <Mike English_web_814> +1 on must-bundle
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[13:17:44] <Sean Turner_web_362> max-bundle-damnit
[13:17:45] <Justin Uberti_web_257> mux-bandle
[13:17:58] <Jonathan Lennox_web_478> mock-bindle
[13:18:04] <Mike English_web_814> :satisfied:
[13:18:20] <Suhas Nandakumar_web_854> Bundle-bundle
[13:18:31] <Jonathan Lennox_web_478> bundle-mcbundleface
[13:18:41] <Sean Turner_web_362> ^ FTW
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[13:18:52] <Mike English_web_814> bundle-bundle-bundle-bundle-mush...
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[13:19:40] <> I  just said that I think some combination of Cullen, Justin, and myself can do the JSEP update
[13:19:59] <Cullen Jennings_web_828> Glad to help on JSEP.
[13:20:24] <Sean Turner_web_362> Many thanks Justin for presenting this so clearly
[13:20:25] <Jonathan Lennox_web_478> I suspect having the existing editors do the revs of each document is gonna make the editing process a lot easier.
[13:20:52] <Justin Uberti_web_257> agreed
[13:21:05] <Justin Uberti_web_257> and hopefully avoiding us being thawed out in 2050
[13:22:19] <Magnus Westerlund_web_259> Do a full update!
[13:22:37] <Spencer Dawkins_web_462> @Justin - that argues strongly for cremation if you author key IETF specifications ... :grinning:
[13:24:31] <Magnus Westerlund_web_259> I would note that doing a write up for the IETF last call with it being an update is not that problematic. The main issue that exist is that the review teams don't see the write up in the IETF last call.
[13:24:38] <Sean Turner_web_362> @Christer - sorry 'bout that
[13:25:19] <Cullen Jennings_web_828> We could note on front of draft that could be removed by RFC Editor which explained what was going on
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[13:30:44] <Justin Uberti_web_257> doesn't insertable streams affect this?
[13:31:06] <Justin Uberti_web_257> (wrt packet content, perhaps?)
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[13:31:40] <> Yeah, probably mkes it worse
[13:31:58] <Sean Turner_web_362> Note Justin in queue
[13:33:23] <Magnus Westerlund_web_259> But can you run SRTP or DTLS in non-encrypted mode? Otherwise you have to counter the encryption to cause a plain text message that matches your intended message. So for WebRTC does this problem exist outside of TURN?
[13:33:27] <> I am +1 on the "short RFC" version
[13:33:48] <> Magnus: yes, because the SRTP encryption is predictable
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[13:35:08] <Jonathan Lennox_web_478> Aren't the fragmented UDP packets marked as fragmented at the IP layer?  NAT ALGs are ignoring IP fragmentation?
[13:37:13] <hta> Yes, they are :-(
[13:37:16] <Justin Uberti_web_257> @Magnus I think this could happen with ICE usernames too
[13:38:29] <Roman Shpount_web_294> This looks like NAT Router security issue not RTCWeb issue
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[13:40:18] <Justin Uberti_web_257> the safe port list probably solves most of these issues. But can this list be applied to ICE candidates?
[13:40:40] <Jonathan Lennox_web_478> That might be an interesting thing to do telemetry on.
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[13:41:02] <hta> I think we're currently applying the blocked port list to ICE candidates, but I have to test to make sure.
[13:44:07] <Sean Turner_web_362> I uploaded the slides
[13:44:12] <Jonathan Lennox_web_478> Only 443
[13:44:17] <hta> and 80
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[13:44:25] <Ted Hardie_web_860> Roman, we cut the cue.
[13:44:25] <Alessandro Amirante_web_378> what about UDP 53?
[13:44:37] <Cullen Jennings_web_828> Just for FYI, we use port 53 in some cases
[13:45:05] <hta> Cullen, Cisco filed a bug on port 80 TURN servers. Can you file one on port 53?
[13:45:08] <Lorenzo Miniero_web_325> Yeah, 80, 443 and 53 are quite common for TURN
[13:45:38] <Alessandro Amirante_web_378> same here. 53 is kinda often let through firewalls
[13:45:44] <Justin Uberti_web_257> libwebrtc tries to avoid ever fragmenting a packet
[13:46:01] <Jonathan Lennox_web_478> To the previous point about SRTP, once you know the keys and can control the plaintext, you can absolutely generate the ciphertext of your choice, as long as you can structure things such that the auth tag doesn't interfere with your attack.
[13:46:04] <Cullen Jennings_web_828> @hta - glad to work with you on whatver needed here
[13:46:16] <Jonathan Lennox_web_478> Insertable streams, or maybe G.711 too.
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[13:47:10] <Justin Uberti_web_257> right. so ICE-TCP + mss tweaking + insertable streams would allow you to make any packet you want
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[13:47:10] <> Thanks Justin for taking the lead on this
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[13:47:14] <Cullen Jennings_web_828> These are best chairs ever - clearly getting stuff done fast
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[13:47:27] <Justin Uberti_web_257> adios all
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[13:48:11] <hta> the attacker would have to extract the symmetric key at the attacker end, so cannot do this during the setup phase
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