IETF
rtcweb@jabber.ietf.org
Thursday, 2 August 2012< ^ >
martin.thomson has set the subject to: RTCWEB WG http://tools.ietf.org/wg/rtcweb/agenda
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[15:57:55] <Lorenzo Miniero> Slide 4: Milestones
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[16:01:22] <Lorenzo Miniero> Slide 2: Note Well
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[16:02:50] <Lorenzo Miniero> Slide 1: DSCP markings for   RTCWeb QoS 
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[16:03:13] <Lorenzo Miniero> Slide 2: What we are talking about …
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[16:04:45] <Lorenzo Miniero> Slide 3: Where this helps …
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[16:05:54] <Alessandro Amirante> Hello everybody! Meetecho is @ www.meetecho.com/ietf84/rtcweb
[16:06:02] <Alessandro Amirante> If you already are in the IETF jabber room, please remember to use a diffrent nickname when you also log into Meetecho.
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[16:06:48] <Muthu Perumal> I am not hearing any audio in Meetecho audio streaming
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[16:07:05] <Muthu Perumal> are others able to hear?
[16:07:11] <Lorenzo Miniero> Slide 4: Setting the DSCP
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[16:07:43] <Lorenzo Miniero> Muthu the RTSP stream should be fine
[16:07:51] <Lorenzo Miniero> or are you using the integrated voip/webrtc?
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[16:08:05] <Alessandro Amirante> Yes, I'm listening to it! It's at rtsp://vancouver.conf.meetecho.com/udrtfev.sdp
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[16:09:18] <Simon Romano> The webrtc stream is ok
[16:09:37] <Lorenzo Miniero> Slide 5: Why three levels?
[16:10:23] <Tobia Castaldi> I'm using the integrated flash based stream and it works fine
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[16:11:05] <Muthu Perumal> integrated flash streaming has started working for me..
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[16:11:18] <Tobia Castaldi> perfect :)
[16:12:14] <eburger> Are you saying that you are having no problem getting our real-time stream without diffserv? How could that be possible? :->
[16:12:22] <Lorenzo Miniero> Slide 6: What this draft is suggesting 
[16:12:26] <Lorenzo Miniero> Slide 4: Setting the DSCP
[16:13:04] <Lorenzo Miniero> Slide 5: Why three levels?
[16:13:08] <Lorenzo Miniero> Slide 6: What this draft is suggesting 
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[16:16:48] <Simon Romano> @eburger: we have traffic-engineered the IETF network :-)
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[16:19:42] <Lorenzo Miniero> Slide 6: What this draft is suggesting 
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[16:20:06] <Michael Tüxen> FreeBSD 9.1 (upcoming) will support it.
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[16:23:30] <Magnus> Michael, did you want that relayed to mic?
[16:24:12] <Michael Tüxen> @Magnus: Yes. "It" means receiving the TOS byte for UDP packet as a CMSG.
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[16:25:14] <wolfgang.beck01> the latency inside the browser/os might be a harder problem than router queues..
[16:25:38] <hta> Michael, Linux supports getting it too (I don't know what call it is, but it's exposed on Google's internal sockets wrappers).
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[16:26:10] <hta> "it" = getting the DSCP setting on a socket, and getting the DSCP that arrived on an incoming packet
[16:26:27] <hta> UDP packet, that is.
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[16:27:59] <Michael Tüxen> hta: I think Linux does it very similar to FreeBSD.
[16:28:25] <Lorenzo Miniero> Slide 7: Next Steps
[16:28:45] <shakansson> "hm"
[16:28:48] <Michael Tüxen> hm
[16:29:11] <Lorenzo Miniero> Presentation stopped
[16:29:27] <hta> personal opinion: just make it a WG draft. Possibly it should go inside another one later, but let's get it as WG-sanctioned the sooner the better.
[16:29:32] <Lorenzo Miniero> Slide 1: RTCWeb Considerations for Mobile Devices
[16:29:44] <Lorenzo Miniero> Slide 2: Scope
[16:31:15] <Lorenzo Miniero> Slide 3: Assumptions
[16:32:27] <Lorenzo Miniero> Slide 4: Cellular Networks
[16:33:50] <shakansson> comment: I think the timers are quite different (much shorter) for LTE
[16:35:19] <hardie> Is that for the mic
[16:35:27] <Lorenzo Miniero> Slide 5: Connectivity to the Calling Site
[16:35:41] <shakansson> perhaps it is enough if it is in the chat
[16:35:49] <hardie> okay
[16:38:21] <Lorenzo Miniero> Slide 6: Media and Data Channels
[16:38:22] <Lorenzo Miniero> Slide 6: Media and Data Channels
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[16:49:00] <Lorenzo Miniero> Slide 7: Interface Switching
[16:49:09] <Lorenzo Miniero> Slide 7: Interface Switching
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[16:51:06] <Lorenzo Miniero> Slide 8: Congestion Avoidance
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[16:55:10] <frodek> There's a W3C "network friendly webapps" task force looking into if there's something that W3C should do in this area (http://www.w3.org/wiki/Network-Friendly-WebApps)
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[16:57:20] <Lorenzo Miniero> Presentation stopped
[16:57:34] <Lorenzo Miniero> Slide 1: RTCWEB WG
[16:57:53] <Lorenzo Miniero> Slide 2: Purpose of the Document
[16:58:35] <Lorenzo Miniero> Slide 3: Caveats
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[16:59:22] <Lorenzo Miniero> Slide 4: Context
[16:59:53] <Lorenzo Miniero> Slide 5: 
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[17:04:58] <Lorenzo Miniero> Slide 6: Accessibility
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[17:10:20] <Richard Barnes> http://geopriv.dreamhosters.com/web911/
[17:10:56] <Tobia Castaldi> Slide 7: Media Requirements
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[17:11:45] <Richard Barnes> ^^^ js prototype of ecrit
[17:11:59] <Tobia Castaldi> Current presenter: Bernard Aboba
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[17:12:00] <Tobia Castaldi> Slide 7: Media Requirements
[17:12:48] <Simon Romano> I keep on losing network connectivity in this over-crowed room :-(. Am I the only one?
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[17:13:12] <Richard Barnes> simon: +1
[17:13:18] <Tobia Castaldi> me too :S
[17:13:24] <Jonathan Lennox> And me
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[17:13:37] <Alessandro> The same here...
[17:13:40] <Simon Romano> I'm getting crazy with the video stream, sorry about that.
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[17:17:11] <Tobia Castaldi> Slide 8: Feedback?
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[17:17:58] <Tobia Castaldi> Slide 3: Caveats
[17:18:05] <Spencer Dawkins> I sent a ticket about this to mailto:tickets@meeting.ietf.org ...
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[17:20:16] <Spencer Dawkins> "this" being connectivity in the room ...
[17:20:34] <Tobia Castaldi> Slide 1: RTCWEB STUN Extensions for  Authentic
[17:20:49] <Tobia Castaldi> Slide 2: Agenda
[17:20:53] <Tobia Castaldi> Slide 3: Problem Statement
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[17:21:22] <Tobia Castaldi> Slide 4: Existing Model: ALG  (Application La
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[17:23:42] <Tobia Castaldi> Slide 5: Proposed Solution
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[17:24:31] <Tobia Castaldi> Slide 6: (Scenario:  WebRTC service in enter
[17:25:41] <Tobia Castaldi> Slide 7: Next steps
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[17:35:47] <Tobia Castaldi> Slide 1: IETF Security Architecture Update
[17:35:52] <Tobia Castaldi> Current presenter: Eric Rescorla
[17:36:14] <Tobia Castaldi> Slide 2: Overview
[17:36:21] <Tobia Castaldi> Slide 3: Reviews
[17:36:32] <Tobia Castaldi> Slide 2: Overview
[17:37:13] <Tobia Castaldi> Slide 3: Reviews
[17:38:04] <Tobia Castaldi> Slide 4: Changes since -02
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[17:40:29] <Tobia Castaldi> Slide 5: IP Location Privacy
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[17:41:13] <Tobia Castaldi> Slide 6: Identity Origin Detection
[17:42:48] <Tobia Castaldi> Slide 7: Proposed change
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[17:43:32] <Tobia Castaldi> Slide 8: Example
[17:43:43] <Tobia Castaldi> Slide 9: Communications Consent Overview
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[17:45:18] <Tobia Castaldi> Slide 10: Consent Freshness
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[17:46:40] <Tobia Castaldi> Slide 11: What method to use?
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[17:49:51] <Tobia Castaldi> Slide 12: Is a MAC needed?
[17:49:58] <Tobia Castaldi> Slide 13: ... But
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[17:58:48] <Tobia Castaldi> Slide 14: Duration of Unwanted Traffic/Timer Setti
[18:00:07] <Tobia Castaldi> Slide 15: Shorter STUN timers
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[18:05:29] <Tobia Castaldi> Slide 16: A related problem: liveness testing
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[18:07:02] <Tobia Castaldi> Slide 17: Combined Consent/Liveness Both checks do
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[18:08:24] <Lorenzo Miniero> for remotes on Meetecho -- sorry, this slide deck seems to be different from the one we had :(
[18:08:37] <Lorenzo Miniero> the current slide is slide 17, "Why not RTCP for liveness"
[18:08:57] <Tobia Castaldi> Slide 17: Combined Consent/Liveness Both checks do
[18:09:12] <Lorenzo Miniero> slide 18 on the beamer, fyi
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[18:16:15] <Tobia Castaldi> Slide 17: Combined Consent/Liveness Both checks do
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[18:17:04] <hta> so was there any decision on the consent issue? I didn't detect one.
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[18:17:21] <Tobia Castaldi> Slide 1: WebRTC: Media Transport and Use of RTP
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[18:17:28] <Tobia Castaldi> Current presenter: Colin Perkins
[18:17:30] <Tobia Castaldi> Slide 2: Status
[18:18:12] <Tobia Castaldi> Slide 3: Choice of RTP Payload Formats
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[18:20:59] <Tobia Castaldi> Slide 4: RTP Session Multiplexing
[18:21:09] <Tobia Castaldi> Slide 5: Generation of the RTCP CNAME
[18:21:29] <Tobia Castaldi> Slide 6: Congestion Control
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[18:22:51] <Tobia Castaldi> Slide 7: Performance Monitoring
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[18:28:51] <Tobia Castaldi> Slide 8: WebRTC API Considerations
[18:29:35] <Tobia Castaldi> Slide 9: RTP Implementation Considerations
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[18:30:22] <Tobia Castaldi> Slide 10: Supported RTP Topologies
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[18:31:16] <Tobia Castaldi> Slide 11: Next Steps
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