IETF
rtcweb@jabber.ietf.org
Tuesday, 26 July 2011< ^ >
martin.thomson has set the subject to: RTCWEB
Room Configuration

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[16:33:26] <rillian> morning!
[16:33:45] <rillian> Is this the proper room for the rtcweb meeting today?
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[16:53:24] <martin.thomson> I am your friendly jabber proxy today. If you want to know what is going on, or you need a mediumm let me know.
[16:53:49] <martin.thomson> Requests for channelling should be prefixed with "mic:"
[16:54:20] <rillian> thanks, martin
[16:54:31] <rillian> no a/v stream yet?
[16:54:42] <tomkrist> rillian: Audio works fine for me
[16:54:53] <martin.thomson> tomkrist, can you share the url?
[16:54:59] <tomkrist> vlc http://ietf81streaming.dnsalias.net/ietf/ietf803.m3u
[16:55:08] <tomkrist> (Well, skip the vlc part :-) )
[16:55:10] <burn> when the meetecho guys show up I'm sure they'll send a link as well
[16:55:22] <martin.thomson> http://tools.ietf.org/wg/rtcweb/agenda
[16:55:25] <martin.thomson> thanks :)
[16:55:47] <rillian> http://www.meetecho.com/ietf81/rtcweb is the meetecho link
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[16:56:00] <tomkrist> Hmm, the audio stopped just now...
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[17:00:30] <rillian> meetecho link is up, but no html5 stream that I can find
[17:00:46] <martin.thomson> welcome to the brave new world
[17:00:47] <Lorenzo Miniero> rillian, the html5 stream is only available for the recordings
[17:01:02] <rillian> oh, the irony :)
[17:01:08] <Lorenzo Miniero> ;)
[17:01:11] <tomkrist> :-)
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[17:02:03] <rillian> still, sad the ietf is still streaming mp3. only a couple of years to redeem ourselves.
[17:02:06] <MeetechoRTCWEB> Presentation stopped
[17:02:08] martin.thomson has set the subject to: RTCWEB WG http://tools.ietf.org/wg/rtcweb/agenda
[17:02:40] <MeetechoRTCWEB> Presentation stopped
[17:02:43] <Lorenzo Miniero> I don't know if our rtsp stream would play in an html5 audio tag
[17:02:47] <Lorenzo Miniero> I doubt that, anyway
[17:02:55] <rillian> Lorenzo Miniero: no, you'd need to provide an http stream
[17:03:07] Gonzalo joins the room
[17:03:29] <martin.thomson> Agenda pt1: Tuesday
[17:03:45] Cary Bran joins the room
[17:04:34] Randell Jesup joins the room
[17:04:35] <martin.thomson> milestone update
[17:04:58] Adam Roach joins the room
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[17:05:19] <martin.thomson> gonzalo at the mic
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[17:05:39] <martin.thomson> current wg status
[17:05:41] <RjS> could people hear gonzalo?
[17:05:51] <rillian> bit quiet
[17:05:52] <Lorenzo Miniero> a bit low
[17:05:55] <martin.thomson> future plan
[17:06:00] <Gonzalo> YEs, I had the feeling the microphone was either off or very low
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[17:06:41] <Gonzalo> OK, it seems to work...
[17:06:58] <tomkrist> Loud and clear
[17:07:16] Roy joins the room
[17:07:25] <martin.thomson> W3C chairs are rising
[17:07:30] sftcd joins the room
[17:07:37] <Randell Jesup> Can hear it, video frozen
[17:08:00] <MeetechoRTCWEB> Slide 1: W3C webrtc chairs report
[17:08:02] <martin.thomson> Status
[17:08:06] <MeetechoRTCWEB> Slide 2: Status
[17:08:31] <martin.thomson> ahh, that's great, I hate having to do the slide change thing
[17:08:57] lixia joins the room
[17:09:02] <Lorenzo Miniero> we're italian, we love helping people :)
[17:09:36] <MeetechoRTCWEB> Slide 3: Next steps
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[17:10:38] <Roy> if the blank part is cut, then perfect! :)
[17:11:12] <MeetechoRTCWEB> Presentation stopped
[17:11:51] <MeetechoRTCWEB> Slide 1: WEBRTC API in Chrome
[17:11:51] <martin.thomson> harald is getting a t-shirt
[17:11:57] <martin.thomson> a "Google" t-shirt
[17:12:18] <MeetechoRTCWEB> Slide 2: Our goal
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[17:13:00] <MeetechoRTCWEB> Slide 3: What we've done so far
[17:13:20] <Gonzalo> I got a few questions about the URI for meetecho. Use this one: http://www.meetecho.com/ietf81/rtcweb
[17:13:34] <MeetechoRTCWEB> Slide 4: Overview of the webrtc.org package
[17:14:30] <MeetechoRTCWEB> Slide 5: Implementation strategy
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[17:15:06] <MeetechoRTCWEB> Slide 6: Issues encountered
[17:15:55] <MeetechoRTCWEB> Slide 7: Status of implementation
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[17:16:04] <MeetechoRTCWEB> Slide 6: Issues encountered
[17:16:08] <MeetechoRTCWEB> Slide 7: Status of implementation
[17:16:14] <MeetechoRTCWEB> Slide 4: Overview of the webrtc.org package
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[17:17:18] <MeetechoRTCWEB> Slide 7: Status of implementation
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[17:17:54] <MeetechoRTCWEB> Presentation stopped
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[17:19:08] <MeetechoRTCWEB> Slide 1: Ongoing Infrastructure Work
[17:20:18] <MeetechoRTCWEB> Slide 2: WebRTC Plans
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[17:22:39] <linyi> don't do what?
[17:22:55] <martin.thomson> DON'T DO SIP -- jdr
[17:23:24] <MeetechoRTCWEB> Presentation stopped
[17:23:39] <martin.thomson> sohel khan at the mic
[17:23:59] Ethan Hugg joins the room
[17:24:36] <rillian> can't hear the question
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[17:24:52] <linyi> how many NATs
[17:25:11] <rillian> in the test of the webrtc code
[17:25:16] <MeetechoRTCWEB> Slide 1: RTC-Web
[17:25:33] <Randell Jesup> Overall, I was very happy with GIPS code as a client (before Google bought them)
[17:25:54] <MeetechoRTCWEB> Slide 2: Design Objectives
[17:26:12] <martin.thomson> cary was given a d006:f00d t-shirt
[17:26:16] <Randell Jesup> It was provided as binary, but my cursory looks at the source it looks good (missing design docs though)
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[17:26:47] <linyi> i have a xxxl size cisco t-shirt
[17:27:07] <linyi> she is pretty
[17:27:18] <MeetechoRTCWEB> Slide 3: Implementation Overview
[17:29:20] <MeetechoRTCWEB> Presentation stopped
[17:29:27] <Ted> Randell: Cary's presentation has more detail in the backup slides: http://www.ietf.org/proceedings/81/slides/rtcweb-6.pdf
[17:29:38] <martin.thomson> jdr at mic
[17:29:48] <MeetechoRTCWEB> Slide 1: ??
[17:30:01] <rillian> so mozilla is writing an extension for firefox, and cisco already has one?
[17:30:37] <martin.thomson> choice is good
[17:30:53] <rillian> it is!
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[17:32:17] <MeetechoRTCWEB> Slide 2: ??
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[17:32:27] <MeetechoRTCWEB> Simple example
[17:32:34] <Gonzalo> http://www.youtube.com/watch?v=cx4rZH7fLpM
[17:32:41] <MeetechoRTCWEB> Slide 3: ??
[17:32:48] <MeetechoRTCWEB> Simple example
[17:32:49] <Gonzalo> This was the URI on the slide
[17:33:14] <MeetechoRTCWEB> Slide 4: ??
[17:33:24] <MeetechoRTCWEB> Toolbox
[17:33:51] <MeetechoRTCWEB> Presentation stopped
[17:34:28] <MeetechoRTCWEB> Slide 1: Use cases and reqs
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[17:34:53] <MeetechoRTCWEB> Slide 2: Intro
[17:35:17] <MeetechoRTCWEB> Slide 3: Simple Video Communication Service
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[17:35:59] <MeetechoRTCWEB> Slide 4: Simple Video Communication Service, acce
[17:36:10] <MeetechoRTCWEB> ...ss change
[17:36:36] <MeetechoRTCWEB> Slide 5: Simple Video Communication Service, QoS
[17:36:50] <MeetechoRTCWEB> Slide 6: Simple Video Communication Service with
[17:36:59] <MeetechoRTCWEB> interop calling
[17:37:02] <MeetechoRTCWEB> Slide 7: Hockey game viewer
[17:37:36] <MeetechoRTCWEB> Slide 8: Multiparty video comm
[17:38:03] <MeetechoRTCWEB> Slide 9: Multiparty on-line game with voice comm
[17:38:48] <MeetechoRTCWEB> Slide 10: Telephony terminal
[17:39:13] <MeetechoRTCWEB> Slide 11: Fedex call
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[17:39:32] <MeetechoRTCWEB> Slide 12: Video conf with server
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[17:40:24] <MeetechoRTCWEB> Slide 13: Proposed new use cases
[17:40:24] <MeetechoRTCWEB> Slide 13: Proposed new use cases
[17:42:27] <MeetechoRTCWEB> Slide 14: Possible documents
[17:42:38] <rillian> oh good, a music use case
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[17:43:08] <MeetechoRTCWEB> :-)
[17:43:29] <MeetechoRTCWEB> Slide 13: Proposed new use cases
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[17:47:27] <martin.thomson> christer at mic
[17:47:31] <rillian> I guess real-time health monitoring data is a case for a *reliable* datagram channel
[17:47:53] <Randell Jesup> yes
[17:48:53] <Randell Jesup> Agree with speaker!
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[17:49:21] <linyi> i also agree with the speaker
[17:49:58] <linyi> but we should not exhaust the discussion on how many use cases we need.
[17:50:07] <Randell Jesup> Agree health care
[17:50:27] <martin.thomson> As I said in geopriv: in privacy, there is plenty of opportunity for regrets
[17:55:31] <martin.thomson> bernard aboba
[17:56:38] <Randell Jesup> VRS is another usecase we should consider
[17:56:51] <Randell Jesup> (Video Relay Service)
[17:57:11] <Randell Jesup> Not sure if it will fall into existing case
[17:57:21] <martin.thomson> mic?
[17:57:33] <Ted> That was Colin, speaking before harald
[17:58:43] <Randell Jesup> Streaming video isn't covered for peer-to-peer
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[18:00:38] <MeetechoRTCWEB> Presentation stopped
[18:00:48] <Randell Jesup> 911 may have other requirements on codec choice, enabling of mics/cameras, etc
[18:01:02] <MeetechoRTCWEB> Slide 1: RTCWEB Architecture
[18:01:08] <MeetechoRTCWEB> Slide 2: What I will talk about
[18:02:06] <Cullen Jennings> I would take a random guess that about 40 people or so said they had the the overview draft
[18:02:07] <MeetechoRTCWEB> Slide 3: What I will not talk about
[18:02:25] <MeetechoRTCWEB> Slide 4: Goal for RTCWEB:
[18:02:34] <Randell Jesup> how many are there?
[18:02:55] <Roy> 150 or more
[18:03:14] <Cullen Jennings> over 150 people on the blue sheets
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[18:04:34] <MeetechoRTCWEB> Slide 5: Architecture in context
[18:05:18] <MeetechoRTCWEB> Slide 6: Architecture layers
[18:05:42] <linyi> it would be helpful to give the link of the presentation at the beginning:)
[18:06:07] <martin.thomson> check the agenda
[18:06:17] <MeetechoRTCWEB> ...or join Meetecho.
[18:06:34] <linyi> http://www.ietf.org/proceedings/81/slides/rtcweb-0.pdf
[18:06:43] <linyi> what is meetecho?
[18:06:46] <linyi> how to join?
[18:07:19] <MeetechoRTCWEB> http://www.meetecho.com/ietf81/rtcweb
[18:07:35] <MeetechoRTCWEB> Slide 7: Security in context
[18:08:00] Roy YUE joins the room
[18:08:40] <MeetechoRTCWEB> Slide 8: Data Transport
[18:09:17] <rillian> MeetechoRTCWEB: I think your slides are ahead-by-one
[18:09:17] Linyi Tian joins the room
[18:09:41] <Gonzalo> The slides seem to be perfectly synchronized now
[18:09:44] Roy YUE leaves the room
[18:09:55] <MeetechoRTCWEB> Slide 9: Data framing and securing
[18:09:57] rlb joins the room
[18:10:17] <Linyi Tian> is this the official tool for ietf?
[18:10:19] <rlb> i'm entering this in the meetecho room; does this forward to jabber?
[18:10:28] <MeetechoRTCWEB> Yep
[18:10:28] <richard.barnes> dear self: apparently so
[18:10:40] <martin.thomson> richard: talk to yourself often?
[18:10:45] <Gonzalo> Yes, meetecho includes a jabber client. The jabber room is the same for everyone
[18:10:46] <Linyi Tian> it's cool.
[18:10:56] <MeetechoRTCWEB> Isn't it?
[18:11:38] <Linyi Tian> if everybody uses this, it will ease the jabber scribe's work. you don't need to mention the slide number:)
[18:11:55] <MeetechoRTCWEB> It's automated, anuhow
[18:11:57] <MeetechoRTCWEB> Slide 10: Data formats
[18:12:17] <MeetechoRTCWEB> I don't like human-based slide scribing...
[18:12:26] <rillian> or maybe the mp3 stream is behind-by-one :(
[18:12:46] <Gonzalo> Yes, the traditional mp3 streams can have a delay of minutes sometimes
[18:12:50] <martin.thomson> the mp3 stream always lags by a little bit, sometimes up to minutes
[18:13:04] <Gonzalo> Mettecho also includes VoIP and video functionality. Give them a try.
[18:13:07] <MeetechoRTCWEB> Slide 11: Connection Management
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[18:13:21] <MeetechoRTCWEB> Try-out the RTSP with VLC...
[18:13:37] <MeetechoRTCWEB> in the Audio/ideo Tab
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[18:14:46] <MeetechoRTCWEB> Slide 12: Presentation, Control, Local functions
[18:15:50] sftcd leaves the room
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[18:16:22] <MeetechoRTCWEB> Slide 13: Summary
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[18:21:42] <Randell Jesup> A user case for a 'huddle' without a central server is a good one
[18:22:07] <Randell Jesup> Not written ip yet.
[18:25:27] <Linyi Tian> The chairs are losing mind:)
[18:26:37] <Linyi Tian> It would be good to use full name in jabber room
[18:27:10] <Ted> Linyi: we lost them a long time ago.
[18:27:38] <Randell Jesup> postpone
[18:27:46] <MeetechoRTCWEB> Presentation stopped
[18:27:47] <Linyi Tian> Ted: you got your mind back:)
[18:27:59] <MeetechoRTCWEB> Slide 1: RTCWeb Security Considerations
[18:28:15] <MeetechoRTCWEB> Slide 2: Consensus (or at least silence) on most
[18:28:26] <MeetechoRTCWEB> ..security issues
[18:30:50] <Linyi Tian> that's right. this is the implmenetation issue.
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[18:34:49] <MeetechoRTCWEB> Slide 3: Scope of user consent
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[18:36:32] <MeetechoRTCWEB> Slide 4: Ad Hoc Calling from Embedded Advertiseme
[18:36:48] <MeetechoRTCWEB> ..nts
[18:37:45] <MeetechoRTCWEB> Slide 5: Threat Impact of Ad-Hoc Calling
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[18:38:28] <MeetechoRTCWEB> Slide 6: Ad Hoc Calling from Embedded Advertiseme
[18:39:33] <MeetechoRTCWEB> Slide 7: OK, so that’s a little better
[18:44:36] <kepeng_li\40jabber.org> Why should I allow others to use my camera? I would always say No.
[18:45:17] <Ted> kepeng—so they can see your face on a video call
[18:45:18] <martin.thomson> did you want someone to be able to see and hear you when you call them?
[18:45:58] <linyi> kepeng definitly wants to see:)
[18:46:04] <MeetechoRTCWEB> Slide 8: What about communications security?
[18:46:07] <linyi> i know him quite well. LOL
[18:46:20] <kepeng_li\40jabber.org> I want to see others, for example, a beautiful girl. :-)
[18:46:34] <linyi> i am confused why in-flow dialog is not allowed.
[18:47:21] <MeetechoRTCWEB> Slide 9: Protecting Against Retrospective Attack
[18:48:19] <Jonathan Lennox> linyi: I agree. Google Hangout always pops up the "check your hair and your microphone" dialog before you join the hangout…wouldn't that be the point at which you'd give approval?
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[18:48:33] <Randell Jesup> end-to-end DH still can be MITM'd - see ZRTP key chaining
[18:48:34] <MeetechoRTCWEB> Slide 10: Protecting Against During-Call Attack
[18:48:35] <Jonathan Lennox> It'd mean that dialog was from the browser not the site
[18:48:50] <martin.thomson> in-flow dialogs don't guarantee actual consent, there's plenty of research to prove that
[18:49:15] <martin.thomson> recall the plenary talk wrt regret
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[18:50:18] <Randell Jesup> ALso tough with forwarding
[18:51:11] <Randell Jesup> Who is talking?
[18:51:17] <Linyi Tian> cullen
[18:51:20] <MeetechoRTCWEB> Slide 11: Allow unencrypted RTP at all?
[18:51:21] <Randell Jesup> WHo answered the question?
[18:51:27] <Ted> This first was Sohel, the second was Cullen
[18:51:27] <Jonathan Lennox> Cullen answered
[18:51:28] <kepeng_li\40jabber.org> Cullen
[18:51:29] <Randell Jesup> Thanks
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[18:52:26] <PHB> CALEA requires the carriers to give the Feds the bits. There is no legal mechanism in the US that requires application providers to enable the carriers to receive readable bits, a big difference
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[18:53:13] <Randell Jesup> phb: however, they require the user be unable the bits are supplied, and if the bits go end-to-end the bits can't be supplied
[18:53:43] <PHB> Randell: I can't parse that statement
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[18:53:55] <martin.thomson> Randell: nor I
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[18:54:11] <Ted> Missing: the user to detect the bits are being supplied to Law enforcement
[18:54:15] <Randell Jesup> There's no way to avoid someone making an RTCWEB gateway to unencrypted, so the arguement in favor of all-encryption is weakened a lot
[18:54:23] <kepeng_li\40jabber.org> If browser wants to support RTP, it should first support SIP, right?
[18:54:52] <Linyi Tian> no
[18:54:56] <Linyi Tian> they are not related
[18:54:57] <Randell Jesup> Grr... app didn't keep up with typing.
[18:55:01] <rillian> kepeng_li: the current web API leave SIP implementation tot he server side
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[18:56:10] <kepeng_li\40jabber.org> thanks, rillian
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[18:56:35] <Randell Jesup> CALEA demands the user be unable to detect tapping. Including by wireshark on the stream and looking at the IPs. TDirect brwoser-to-browser connections can be monitored, so if it usually goes directly and suddenly is relayed through fcc.gov :-) it's obvious it wa s tapped
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[18:57:37] <Randell Jesup> To rosenberg: there's no way to avoid it getting unencrypted on the far side of a gateway.
[18:58:02] <Randell Jesup> Exactly
[18:58:15] <Randell Jesup> agree with speaker
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[18:59:33] <Randell Jesup> There is an additional cost in delay and network hardware
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[19:00:26] <Linyi Tian> in most of the cases encryption maybe needed if camera and mic are open. so Ted is right
[19:00:46] <Randell Jesup> THis isn't to say I can't be persuaded all-encryption is the way to go.
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[19:01:25] <Randell Jesup> ICE isn't huge in practice here I think.
[19:01:50] <Randell Jesup> Some good points there
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[19:02:19] <kepeng_li\40jabber.org> I guess we don't have time to discuss the rest of 4 drafts. Will we continute in Session II about the drafts?
[19:02:40] <Randell Jesup> Exactly some of my poionts
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[19:03:05] <Randell Jesup> I'm on the fence
[19:03:10] <PHB> @Randall So comcast has a problem when people use these programs. But Congress has not placed a legal requirement on the program developers.
[19:03:50] <MeetechoRTCWEB> Goodbye everybody! See you soon on Meetecho for next RTCWEB session on Thursday (1pm) :-)
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[19:04:30] <Randell Jesup> phb: so far. However any network provider that touched the PSTN is covered. Likely that will expand in the next few years
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[19:04:53] <Tom Kristensen> Ciao
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[19:04:54] <Randell Jesup> Skype is covered, for example
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[19:05:15] <Randell Jesup> BTW, in theory that includes video streams too
[19:05:26] <Randell Jesup> Though some providers ignore that
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[19:05:56] <MeetechoRTCWEB> Presentation stopped
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[19:06:52] <Randell Jesup> This is the problem in theory of the application providers; they may be forced to proxy all streams for example
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[19:07:08] <tomkrist> Ciao
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[19:08:02] <Randell Jesup> See you all tomorrow...
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