Web Real-Time Communications Working Group Wiki - User contributions [en]

February 6 - February 7 2013

Shkansso2 — Thu, 07 Feb 2013 15:50:07 GMT

Shkansso2: /* Presentations */

== Logistics ==
This meeeting is preceded by the [http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013 Media Capture TF meeting].
=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

We'll use WebEx. Details at http://lists.w3.org/Archives/Public/public-webrtc/2013Feb/0006.html

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): HTA
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

== Reading list ==
* [http://dev.w3.org/2011/webrtc/editor/webrtc.html Latest WebRTC API spec]
* [http://dev.w3.org/2011/webrtc/editor/getusermedia.html Latest Media Capture and Streams spec]

== Presentations ==
Call flows - Adam: http://www.w3.org/2011/04/webrtc/wiki/images/3/3c/WebRTC-call-flows.pdf

http://www.w3.org/2011/04/webrtc/wiki/images/8/89/WebRTC-call-flows.pptx

Data channel - Randell: http://www.w3.org/2011/04/webrtc/wiki/images/d/db/DataChannel_JS_API.pdf

Aspect Ratio - Justin: http://www.w3.org/wiki/images/7/76/Aspect_Ratio.pdf

Stats - Harald: http://www.w3.org/2011/04/webrtc/wiki/images/b/bf/RTCWEB_Interim_Stats.pdf

File:RTCWEB Interim Stats.pdf

Shkansso2 — Thu, 07 Feb 2013 15:49:37 GMT

Shkansso2:

February 6 - February 7 2013

Shkansso2 — Thu, 07 Feb 2013 13:53:22 GMT

Shkansso2: /* Presentations */

File:DataChannel JS API.pdf

Shkansso2 — Thu, 07 Feb 2013 13:52:44 GMT

Shkansso2: uploaded a new version of "File:DataChannel JS API.pdf"

February 6 - February 7 2013

Shkansso2 — Thu, 07 Feb 2013 13:17:01 GMT

Shkansso2: /* Presentations */

== Logistics ==
This meeeting is preceded by the [http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013 Media Capture TF meeting].
=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

We'll use WebEx. Details at http://lists.w3.org/Archives/Public/public-webrtc/2013Feb/0006.html

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): HTA
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

== Reading list ==
* [http://dev.w3.org/2011/webrtc/editor/webrtc.html Latest WebRTC API spec]
* [http://dev.w3.org/2011/webrtc/editor/getusermedia.html Latest Media Capture and Streams spec]

== Presentations ==
Call flows - Adam: http://www.w3.org/2011/04/webrtc/wiki/images/3/3c/WebRTC-call-flows.pdf

http://www.w3.org/2011/04/webrtc/wiki/images/8/89/WebRTC-call-flows.pptx

Data channel - Randell: http://www.w3.org/2011/04/webrtc/wiki/images/d/db/DataChannel_JS_API.pdf

Aspect Ration - Justin: http://www.w3.org/wiki/images/7/76/Aspect_Ratio.pdf

February 6 - February 7 2013

Shkansso2 — Wed, 06 Feb 2013 17:00:10 GMT

Shkansso2: /* Presentations */

File:WebRTC-call-flows.pptx

Shkansso2 — Wed, 06 Feb 2013 16:59:28 GMT

Shkansso2:

February 6 - February 7 2013

Shkansso2 — Wed, 06 Feb 2013 14:06:38 GMT

Shkansso2: /* Presentations */

== Logistics ==
This meeeting is preceded by the [http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013 Media Capture TF meeting].
=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

We'll use WebEx. Details at http://lists.w3.org/Archives/Public/public-webrtc/2013Feb/0006.html

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): HTA
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

== Reading list ==
* [http://dev.w3.org/2011/webrtc/editor/webrtc.html Latest WebRTC API spec]
* [http://dev.w3.org/2011/webrtc/editor/getusermedia.html Latest Media Capture and Streams spec]

== Presentations ==
Call flows - Adam: http://www.w3.org/2011/04/webrtc/wiki/images/3/3c/WebRTC-call-flows.pdf

Data channel - Randell: http://www.w3.org/2011/04/webrtc/wiki/images/d/db/DataChannel_JS_API.pdf

February 6 - February 7 2013

Shkansso2 — Wed, 06 Feb 2013 14:06:27 GMT

Shkansso2: /* Presentations */

== Logistics ==
This meeeting is preceded by the [http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013 Media Capture TF meeting].
=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

We'll use WebEx. Details at http://lists.w3.org/Archives/Public/public-webrtc/2013Feb/0006.html

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): HTA
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

== Reading list ==
* [http://dev.w3.org/2011/webrtc/editor/webrtc.html Latest WebRTC API spec]
* [http://dev.w3.org/2011/webrtc/editor/getusermedia.html Latest Media Capture and Streams spec]

== Presentations ==
Call flows - Adam: http://www.w3.org/2011/04/webrtc/wiki/images/3/3c/WebRTC-call-flows.pdf
Data channel - Randell: http://www.w3.org/2011/04/webrtc/wiki/images/d/db/DataChannel_JS_API.pdf

File:DataChannel JS API.pdf

Shkansso2 — Wed, 06 Feb 2013 14:06:00 GMT

Shkansso2:

February 6 - February 7 2013

Shkansso2 — Wed, 06 Feb 2013 14:03:12 GMT

Shkansso2: /* Presentations */

File:WebRTC-call-flows.pdf

Shkansso2 — Wed, 06 Feb 2013 14:02:59 GMT

Shkansso2:

February 6 - February 7 2013

Shkansso2 — Wed, 06 Feb 2013 13:18:03 GMT

Shkansso2: /* Remote participation */

February 6 - February 7 2013

Shkansso2 — Wed, 30 Jan 2013 09:23:04 GMT

Shkansso2:

== Logistics ==
This meeeting is preceded by the [http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013 Media Capture TF meeting].
=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

TBA

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): HTA
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

== Reading list ==
* [http://dev.w3.org/2011/webrtc/editor/webrtc.html Latest WebRTC API spec]
* [http://dev.w3.org/2011/webrtc/editor/getusermedia.html Latest Media Capture and Streams spec]

== Presentations ==

February 6 - February 7 2013

Shkansso2 — Wed, 30 Jan 2013 09:22:39 GMT

Shkansso2:

== Logistics ==
This meeeting is preceded by the [http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013 Media Capture TF meeting].
=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

TBA

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): HTA
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

== Reading list ==
* [http://dev.w3.org/2011/webrtc/editor/webrtc.html Latest WebRTC API spec]
* [http://dev.w3.org/2011/webrtc/editor/getusermedia.html Latest Media Capture and Streams spec]
* [https://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/proposals/SettingsAPI_proposal_v6.html Proposal: Media Capture and Streams Settings API]

== Presentations ==

February 6 - February 7 2013

Shkansso2 — Tue, 29 Jan 2013 08:49:00 GMT

Shkansso2:

== Logistics ==

=== IRC ===
The usual: #webrtc at irc.w3.org

=== Remote participation ===

TBA

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): Hta
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

February 6 - February 7 2013

Shkansso2 — Tue, 29 Jan 2013 08:48:33 GMT

Shkansso2:

== Logistics ==

=== IRC ===
#webrtc at irc.w3.org

=== Remote participation ===

TBA

== Agenda ==

=== Feb 6th - 1 hour ===
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

=== Feb 7th - 4 hours ===

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): Hta
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

February 6 - February 7 2013

Shkansso2 — Tue, 29 Jan 2013 08:33:23 GMT

Shkansso2:

Agenda:
Feb 6th - 1 hour
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

Feb 7th - 4 hours

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Stats (1h): Hta
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

February 6 - February 7 2013

Shkansso2 — Tue, 29 Jan 2013 07:34:24 GMT

Shkansso2:

Agenda:
Feb 6th - 1 hour
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

Feb 7th - 4 hours

* MediaStream and MediaStreamTrack - SDP interface (1.5h): (Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Rollback (hope this is just API, and settled)
* Stats (1h): Hta
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

February 6 - February 7 2013

Shkansso2 — Tue, 29 Jan 2013 07:33:53 GMT

Shkansso2: Created page with "Agenda: Feb 6th - 1 hour * Call flow: Adam R ** Walk through examples in spec ** Note disagreements on “what happens here” ** Note questions about “what happens here if..…"

Agenda:
Feb 6th - 1 hour
* Call flow: Adam R
** Walk through examples in spec
** Note disagreements on “what happens here”
** Note questions about “what happens here if....”
** Don’t make decisions - list questions

Feb 7th - 4 hours

* MediaStream and MediaStreamTrack - SDP interface (1.5h):
(Based on the conclusions from rtcweb/mmusic)
** What id’s/labels needs to be signaled
** How to handle rejection (support now? How?)
* Error handling (0.5h):
** Sticking to current principles
** Specific errors and additional error data
* Data channel (0.5h): Randell
** Verify that we pass all info needed for SCTP setup, and vice versa
* Rollback (hope this is just API, and settled)
* Stats (1h): Hta
** Stats needed, whether current are well defined
** Where should stats specs live? (IANA registry, IETF doc, W3C doc, all of the above?)
* Next steps, open issues list (0.5h): Chairs

Main Page

Shkansso2 — Tue, 29 Jan 2013 07:30:51 GMT

Shkansso2: /* Meetings */

= Web Real-Time Communications Working Group =

This is the Wiki of the W3C Real-Time Communications Working Group. The wiki is the main point of entry for participants of the group as well as for people interesting in the day-to-day life of the group. It is used to collect drafts, ideas and logistics information. Higher-level information about the Web Real-Time Communications Workin Group is available on the [http://www.w3.org/2011/04/webrtc/ main home page] of the group.

== Meetings ==
* [[February 6 - February 7 2013]] in Boston, USA
* [[October 29 - October 30 2012]] in Lyon, France
* [[June 11 2012]] in Stockholm, Sweden
* [[February 1 2012]] Feb 1st, Mountain View, CA
* [[October 31 - November 1 2011]] in Santa Clara, California, USA
* [[July 23 2011]] in Quebec City, Canada

In addition, there are teleconferences at varying intervals between these. Background materials can sometimes be found on the [[Teleconferences]] page.

Minutes from meetings and teleconferences are linked from the [http://www.w3.org/2011/04/webrtc/ official WG page].

== On-going work ==

=== API document ===
[http://dev.w3.org/2011/webrtc/editor/webrtc.html Editors' draft]

=== Use cases and requirements ===

The [http://dev.w3.org/2011/webrtc/editor/webrtc_reqs.html Editor's draft of Use cases and requirements] currently merely refers to the [http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/ equivalent IETF document]. In the IETF document use cases and requirements are listed. New use cases that have been proposed but not yet decided are also listed (with links).

There are however some items that should be discussed more within the context of current use cases and requirements (mostly taken from [http://www.ietf.org/mail-archive/web/rtcweb/current/msg01177.html this mail]):

* QoS. Unclear to a large extent at this time. Skip for first version?
* The use case (4.2.5) of using SIP between service providers must be sorted out to know what "SIP" really means and furthermore if all webrtc functionality would be available or if there are certain things you can't do. Also F24 should probably be updated, along with a new API requirement (text proposed from the mail "Web API MUST NOT prevent two webapps that happen to choose to peer with SIP from peering."). Also, the current A18 is vague and it can be discussed how much this is an API question
* What of more "advanced" audio media processing should remain in webrtc, and what should be moved to the Audio WG? Will have implications on F13 and F15 and A14.
* A more advanced IVR use case, or at least more detailed should perhaps replace the text in 4.3.2
* Codec requirements.....
* Section 7.2 (Security): should these requirements be moved into the F-req part?
* More work needed on data channels (unreliable, reliable, file-transfer) in terms of actual use and of requirements.

== Useful Resources ==

=== Mailing-lists ===
* [http://lists.w3.org/Archives/Public/public-webrtc/ Archives of the public-webrtc@w3.org mailing-list] are publicly available. This mailing-list should be used for all technical discussions within the group.
* [http://lists.w3.org/Archives/Member/member-webrtc/ Archives of the member-webrtc@w3.org mailing-list] are available to W3C Members only. Use of this second list should be restricted to administrative purposes (e.g. sending regrets or F2F logistics)

=== Tracker ===
* [http://www.w3.org/2011/04/webrtc/track/ Tracker] is used to track actions within the group. We're only using the ACTION section.
* [https://www.w3.org/Bugs/Public/describecomponents.cgi?product=WebRTC%20Working%20Group Buganizer] is available for tracking issues and bugs.

The [[Tracker usage guidelines]] (Work In Progress) gives details on how and when to use these tools.

=== IRC ===
The group uses IRC to take minutes and exchange info during calls and face-to-face meetings:

* IRC info: [irc://irc.w3.org:6665#webrtc '''#webrtc''' on irc.w3.org, port 6665]
* You may use the [http://irc.w3.org/ W3C Web client], specifying a username and channel #webrtc
* You may find more info on [http://www.w3.org/Project/IRC/ IRC usage at W3C]

=== Other Useful W3C Resources ===
* [http://www.w3.org/2005/10/Process-20051014/ World Wide Web Consortium Process Document] 14 October 2005
* [http://www.w3.org/Guide/ Art of Consensus Guide] (W3C member confidential)
* [http://www.w3.org/Project/IRC/ Notes on IRC at W3C]
* [http://www.w3.org/2001/12/zakim-irc-bot.html The Zakim IRC Teleconference Agent]
* [http://www.w3.org/Guide/1998/08/TeleconferenceHowTo.htm Setting up a telco using Zakim]
* [http://dev.w3.org/2002/scribe/scribedoc.htm#Quick_Start_Guide Quick Start Guide to taking minutes]

Teleconferences

Shkansso2 — Thu, 10 Jan 2013 11:38:31 GMT

Shkansso2: /* Documents */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Jan 10 2013 ===
Time: Jan 10th, 5pm - 6:30pm CET

Zakim: Thursday, 10 January 16:00-17:30 UTC (11:00am-12:30pm Boston local) Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")

IRC: irc.w3.org, #webrtc

==== Agenda ====
(first proposal)

* Welcome, scribe
* approve minutes
* Tech:
** Stats
** Error Handling
** Priority API

* Organizational
** Schedule
** Dependencies on RTCWEB deliverables
** Use of f2f time

==== Documents ====
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [[File:20130110error_handling.pdf‎]] - Error handling - Dan
* [[File:Priority (and some other stuff).pdf‎]] - Priority and some other stuff - Stefan

=== Thu, Dec 13 2012 ===
CHANGED CONFERENCE CODE: 26631#

==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - Presentation: States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Presentation: Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0034/DTMFAPI-v3.pdf Document: DTMF API v3]
* [[File:20130110error handling.pdf]] - Presentation: Error handling - Dan Burnett

File:Priority (and some other stuff).pdf

Shkansso2 — Thu, 10 Jan 2013 11:38:06 GMT

Shkansso2: uploaded a new version of "File:Priority (and some other stuff).pdf"

Teleconferences

Shkansso2 — Thu, 10 Jan 2013 11:22:31 GMT

Shkansso2: /* Documents */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Jan 10 2013 ===
Time: Jan 10th, 5pm - 6:30pm CET

Zakim: Thursday, 10 January 16:00-17:30 UTC (11:00am-12:30pm Boston local) Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")

IRC: irc.w3.org, #webrtc

==== Agenda ====
(first proposal)

* Welcome, scribe
* approve minutes
* Tech:
** Stats
** Error Handling
** Priority API

* Organizational
** Schedule
** Dependencies on RTCWEB deliverables
** Use of f2f time

==== Documents ====
* [[File:20130110error_handling.pdf‎]] - Error handling - Dan
* [[File:Priority (and some other stuff).pdf‎]] - Priority and some other stuff - Stefan
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand

=== Thu, Dec 13 2012 ===
CHANGED CONFERENCE CODE: 26631#

==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - Presentation: States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Presentation: Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0034/DTMFAPI-v3.pdf Document: DTMF API v3]
* [[File:20130110error handling.pdf]] - Presentation: Error handling - Dan Burnett

File:Priority (and some other stuff).pdf

Shkansso2 — Thu, 10 Jan 2013 09:56:43 GMT

Shkansso2: uploaded a new version of "File:Priority (and some other stuff).pdf"

Teleconferences

Shkansso2 — Wed, 09 Jan 2013 17:03:26 GMT

Shkansso2: /* Thu, Jan 10 2013 */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Jan 10 2013 ===
Time: Jan 10th, 5pm - 6:30pm CET

Zakim: Thursday, 10 January 16:00-17:30 UTC (11:00am-12:30pm Boston local) Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")

IRC: irc.w3.org, #webrtc

==== Agenda ====
(first proposal)

* Welcome, scribe
* approve minutes
* Tech:
** Stats
** Error Handling
** Priority API

* Organizational
** Schedule
** Dependencies on RTCWEB deliverables
** Use of f2f time

==== Documents ====
* [[File:Priority (and some other stuff).pdf‎]] - Presentation: Priority and some other stuff - Stefan
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand

=== Thu, Dec 13 2012 ===
CHANGED CONFERENCE CODE: 26631#

==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - Presentation: States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Presentation: Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0034/DTMFAPI-v3.pdf Document: DTMF API v3]

Teleconferences

Shkansso2 — Wed, 09 Jan 2013 16:57:46 GMT

Shkansso2: /* Documents */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Jan 10 2013 ===

==== Agenda ====
(first proposal)

* Welcome, scribe
* approve minutes
* Tech:
** Stats
** Error Handling
** Priority API

* Organizational
** Schedule
** Dependencies on RTCWEB deliverables
** Use of f2f time

==== Documents ====
* [[File:Priority (and some other stuff).pdf‎]] - Presentation: Priority and some other stuff - Stefan
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand

=== Thu, Dec 13 2012 ===
CHANGED CONFERENCE CODE: 26631#

==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - Presentation: States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Presentation: Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0034/DTMFAPI-v3.pdf Document: DTMF API v3]

Teleconferences

Shkansso2 — Tue, 08 Jan 2013 09:34:42 GMT

Shkansso2: /* Documents */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Jan 10 2013 ===

==== Agenda ====
(first proposal)

* Welcome, scribe
* approve minutes
* Tech:
** Stats
** Error Handling
** Priority API

* Organizational
** Schedule
** Dependencies on RTCWEB deliverables
** Use of f2f time

==== Documents ====
* [[File:Priority (and some other stuff).pdf‎]] - Presentation: Priority and some other stuff - Stefan

=== Thu, Dec 13 2012 ===
CHANGED CONFERENCE CODE: 26631#

==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - Presentation: States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Presentation: Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0034/DTMFAPI-v3.pdf Document: DTMF API v3]

File:Priority (and some other stuff).pdf

Shkansso2 — Tue, 08 Jan 2013 09:33:20 GMT

Shkansso2:

Teleconferences

Shkansso2 — Mon, 07 Jan 2013 14:46:22 GMT

Shkansso2: /* WEBRTC Teleconferences */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Jan 10 2013 ===

==== Agenda ====
(first proposal)

* Welcome, scribe
* approve minutes
* Tech:
** Stats
** Error Handling
** Priority API

* Organizational
** Schedule
** Dependencies on RTCWEB deliverables
** Use of f2f time

==== Documents ====
=== Thu, Dec 13 2012 ===
CHANGED CONFERENCE CODE: 26631#

==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - Presentation: States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Presentation: Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf Document: Stats API proposal v2] - Harald Alvestrand
* [http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0034/DTMFAPI-v3.pdf Document: DTMF API v3]

Teleconferences

Shkansso2 — Thu, 13 Dec 2012 11:51:47 GMT

Shkansso2: /* Documents */

== WEBRTC Teleconferences ==
This page is intended to collect the documents for the WEBRTC WG's teleconferences.

=== Thu, Dec 13 2012 ===
==== Agenda ====

Tech:
* States (Justin)
** Rollback
* DTMF (Harald)
* Stats (Harald)
* pool model (Martin)
* error handling (Dan)

Org
* Schedule
* Use of f2f

==== Documents ====
* [[File:Telco-2012-12-13-States.pdf‎]] - States and the Rollback operation - Justin Uberti
* [[File:Telco-2012-12-13-Candidate warm up.pptx]] - Candidate warmup - Martin Thomson (as PDF: [[File:Telco-2012-12-13-Candidate_warm_up-1.pdf‎]])
* http://lists.w3.org/Archives/Public/public-webrtc/2012Dec/att-0013/RTCWEBStatsv2.pdf - Stats API proposal v2 - Harald Alvestrand

Teleconferences

Shkansso2 — Thu, 13 Dec 2012 11:51:15 GMT

Shkansso2: /* Documents */

File:TPAC 2012 - DTMF.pptx

Shkansso2 — Tue, 30 Oct 2012 08:50:43 GMT

Shkansso2:

October 29 - October 30 2012

Shkansso2 — Tue, 30 Oct 2012 08:50:24 GMT

Shkansso2: /* Presentations */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (PeerConnection related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* States - Cullen, Justin (1530-1630)
** PeerConnection
** ICE agent
** What is exposed to application?

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* Identity handling - EKR (0830-0930 - 1 hour)

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Presentations ==
* [[media:MoreAPIs.pdf]] Stefan's slides - very first, very rough draft (and there are trailing slides after 15 that should have been removed)
* [[media:TPAC_2012_WEBRTC_hta.pdf]] - HTA's slides on various points
* [[media:TPAC_2012_WebRTC_error.pdf]] - Anant's slides on error handling
* [[media:RTCWeb-SDP-API_v6.pdf‎]] - Cullen/Martin slides on SDP
* [[media:WebRTC_CallFlow_v2.pdf]] - Cullen slides on Call Flows
* [[media:WebRTC_States_v2.pdf‎ ]] - Cullen/Justin slides on States
* [[media:idp-issues.pdf]] - EKR slides on Identity
* [[media:TPAC 2012 - DTMF.pptx]] - Justin's slides on DTMF

== Minutes ==
(anything below this is added after the meeting)

File:MoreAPIs.pdf

Shkansso2 — Tue, 30 Oct 2012 06:02:36 GMT

Shkansso2: uploaded a new version of "File:MoreAPIs.pdf"

October 29 - October 30 2012

Shkansso2 — Mon, 29 Oct 2012 09:15:48 GMT

Shkansso2: /* Agenda */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (PeerConnection related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* States - Cullen, Justin (1530-1630)
** PeerConnection
** ICE agent
** What is exposed to application?

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* Identity handling - EKR (0830-0930 - 1 hour)

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Presentations ==
* [[media:MoreAPIs.pdf]] Stefan's slides - very first, very rough draft (and there are trailing slides after 15 that should have been removed)
* [[media:TPAC_2012_WEBRTC_hta.pdf]] - HTA's slides on various points

* [[media:RTCWeb-SDP-API_v6.pdf‎]] - Cullen/Martin slides on SDP
* [[media:WebRTC_CallFlow_v2.pdf]] - Cullen slides on Call Flows
* [[media:WebRTC_States_v2.pdf‎ ]] - Cullen/Justin slides on States

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Mon, 29 Oct 2012 06:45:52 GMT

Shkansso2: /* Agenda */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (PeerConnection related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* Identity handling - EKR (1530-1630 - 1 hour)

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* States - Cullen, Justin (0830-0930)
** PeerConnection
** ICE agent
** What is exposed to application?

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Presentations ==
very first, very rough draft (and there are trailing slides after 15 that should have been removed) [[media:MoreAPIs.pdf]]

== Minutes ==
(anything below this is added after the meeting)

File:MoreAPIs.pdf

Shkansso2 — Mon, 29 Oct 2012 06:29:19 GMT

Shkansso2: uploaded a new version of "File:MoreAPIs.pdf"

October 29 - October 30 2012

Shkansso2 — Fri, 26 Oct 2012 13:19:46 GMT

Shkansso2: /* Presentations */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* Identity handling - EKR (1530-1630 - 1 hour)

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* States - Cullen, Justin (0830-0930)
** PeerConnection
** ICE agent
** What is exposed to application?

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Presentations ==
very first, very rough draft (and there are trailing slides after 15 that should have been removed) [[media:MoreAPIs.pdf]]

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Fri, 26 Oct 2012 12:59:07 GMT

Shkansso2: /* Presentations */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* Identity handling - EKR (1530-1630 - 1 hour)

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* States - Cullen, Justin (0830-0930)
** PeerConnection
** ICE agent
** What is exposed to application?

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Presentations ==
[[media:MoreAPIs.pdf]]

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Fri, 26 Oct 2012 12:56:43 GMT

Shkansso2: /* Expected attendants */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* Identity handling - EKR (1530-1630 - 1 hour)

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* States - Cullen, Justin (0830-0930)
** PeerConnection
** ICE agent
** What is exposed to application?

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Presentations ==
[[MoreAPIs.pdf]]

== Minutes ==
(anything below this is added after the meeting)

File:MoreAPIs.pdf

Shkansso2 — Fri, 26 Oct 2012 12:55:47 GMT

Shkansso2:

October 29 - October 30 2012

Shkansso2 — Fri, 26 Oct 2012 09:14:00 GMT

Shkansso2: /* Remote participation */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

UW_Web RTC
Monday, 29 Oct - Tuesday, 30 Oct 07:00-18:00 UTC (3:00am-2:00pm Boston local)
Zakim Bridge +1.617.761.6200, conference 9782 ("WRTC")
10 participants

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff (15 min)

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan (0845-1030 - 1h 45 min)
** Sender side (all per track)
*** Priority
*** bandwidth
*** width/height
*** agc toggle switch
*** bw/cong. feedback
*** What can we do with constraints, what needs explicit API surface?
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen (1045-1200 1h 15 min)
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?
*** What does an application tell the UA when a call is rejected?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status (1300-1330)

* General error handling principles - Anant (1330-1400)
** Presentation of principles
** Application to specific points in the API

* Call flows - Cullen’s sample flows - Cullen (1400-1500 - 1 hour)
** Walk through a flow or two, verify that we understand what the transitions mean

* Coffee break (1500-1530)

* Identity handling - EKR (1530-1630 - 1 hour)

* Stats - Harald (1630-1730 - 1 hour)
** Conclude stats model
*** Naming of stats
*** Grouping of stats objects
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Finish for the day (1800)

=== Tuesday morning 0830-1200 ===

* States - Cullen, Justin (0830-0930)
** PeerConnection
** ICE agent
** What is exposed to application?

* API functionality for remove stream - Harald (0930-0945 - 15 min)
** Empty indices in local/remoteStreams or not?

* Conclude DTMF (we hope) - Harald, Justin (0945-1000 - 15 min)

* Any Other Business (1000-1200)
** Items can be raised here from other slots at need

* End of WebRTC WG meeting (afternoon is Media Capture)

=== Media Capture TF meeting ===
See [http://www.w3.org/wiki/Oct_30_2012]

== Expected attendants ==

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Mon, 22 Oct 2012 12:52:10 GMT

Shkansso2: /* Monday afternoon 1300-1800 */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

TBD

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan
** Sender side (all per track)
*** Priority
*** bw
*** width/height
*** agc toggle switch
*** bw/cong. feedback
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* Implementation/testing status

* General error handling principles - Anant

* Call flows - Cullen’s sample flows - Cullen
** Walk through a flow or two, verify that we understand what the transitions mean

* Stats - Harald
** Conclude stats model
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Conclude DTMF (we hope) - Harald, Justin

* Finish for the day

=== Tuesday morning 0830-1200 ===

* API functionality for remove stream - Harald
** Empty indices in local/remoteStreams or not?

* States - Cullen, Justin
** PeerConnection
** ICE agent
** What is exposed to application?

* End of WebRTC WG meeting (afternoon is Media Capture)

== Expected attendants ==

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Mon, 22 Oct 2012 10:50:41 GMT

Shkansso2: /* Agenda */

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

TBD

== Agenda ==

=== Monday morning 0830--1200 ===

* welcome and stuff

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan
** Sender side (all per track)
*** Priority
*** bw
*** width/height
*** agc toggle switch
*** bw/cong. feedback
** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)
** Not certain what side
*** Control of AEC (echo cancellation)
** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?
** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?

* Lunch 1200-1300

=== Monday afternoon 1300-1800 ===

* General error handling principles - Anant

* Call flows - Cullen’s sample flows - Cullen
** Walk through a flow or two, verify that we understand what the transitions mean

* Stats - Harald
** Conclude stats model
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Conclude DTMF (we hope) - Harald, Justin

* Finish for the day

=== Tuesday morning 0830-1200 ===

* API functionality for remove stream - Harald
** Empty indices in local/remoteStreams or not?

* States - Cullen, Justin
** PeerConnection
** ICE agent
** What is exposed to application?

* End of WebRTC WG meeting (afternoon is Media Capture)

== Expected attendants ==

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Mon, 22 Oct 2012 10:47:51 GMT

Shkansso2:

== Logistics ==

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

TBD

== Agenda ==
Monday morning 0830--1200
=========================
* welcome and stuff

* API functionality (MediaStream related) that has been discussed but is not currently supported, what of this should we support in v1? - Stefan

** Sender side (all per track)
*** Priority
*** bw
*** width/height
*** agc toggle switch
*** bw/cong. feedback

** Receiver side
*** reject certain or all tracks in offer
**** should app be able to?
*** inform the sender of used / useful width/height
*** tell the sender that a stream/track is not played (paused/unattached/ended)

** Not certain what side
*** Control of AEC (echo cancellation)

** Agree on what of the above we need
*** What in v1; what we can move to v2 but “make room for” in v1; what we can’t see need for
*** One time, or possible to change?

** Discussion leads to requirements to IETF on SDP and RTCP capabilities

* Coffee break 1030-1045

* SDP handling - Martin T, Cullen
** How long is an SDP created with createOffer/Answer valid?
** Rollback funtionality
*** at setLocal(offer) at offerer
*** at setRemote(offer) at answerer
*** at setLocal(answer) at answerer
*** at setRemote(answer) at offerer
*** What does an application do to recover when the UA rejects the set* call?

* Lunch 1200-1300

Monday afternoon 1300-1800
==========================
* General error handling principles - Anant

* Call flows - Cullen’s sample flows - Cullen
** Walk through a flow or two, verify that we understand what the transitions mean

* Stats - Harald
** Conclude stats model
** Specific stats for
*** Transports
*** ICE addresses / address pairs
*** ???

* Conclude DTMF (we hope) - Harald, Justin

* Finish for the day

Tuesday morning 0830-1200
=========================
* API functionality for remove stream - Harald
** Empty indices in local/remoteStreams or not?

* States - Cullen, Justin
** PeerConnection
** ICE agent
** What is exposed to application?

* End of WebRTC WG meeting (afternoon is Media Capture)

== Expected attendants ==

== Minutes ==
(anything below this is added after the meeting)

October 29 - October 30 2012

Shkansso2 — Mon, 22 Oct 2012 10:42:34 GMT

Shkansso2: Created page with "=== title ==="

=== title ===

Main Page

Shkansso2 — Mon, 22 Oct 2012 08:32:19 GMT

Shkansso2: /* Meetings */

= Web Real-Time Communications Working Group =

This is the Wiki of the W3C Real-Time Communications Working Group. The wiki is the main point of entry for participants of the group as well as for people interesting in the day-to-day life of the group. It is used to collect drafts, ideas and logistics information. Higher-level information about the Web Real-Time Communications Workin Group is available on the [http://www.w3.org/2011/04/webrtc/ main home page] of the group.

== Meetings ==
* [[October 29 - October 30 2012]] in Lyon, France
* [[June 11 2012]] in Stockholm, Sweden
* [[February 1 2012]] Feb 1st, Mountain View, CA
* [[October 31 - November 1 2011]] in Santa Clara, California, USA
* [[July 23 2011]] in Quebec City, Canada

In addition, there are teleconferences at varying intervals between these.

Minutes from meetings and teleconferences are linked from the [http://www.w3.org/2011/04/webrtc/ official WG page].

== On-going work ==

=== API document ===
[http://dev.w3.org/2011/webrtc/editor/webrtc.html Editors' draft]

=== Use cases and requirements ===

The [http://dev.w3.org/2011/webrtc/editor/webrtc_reqs.html Editor's draft of Use cases and requirements] currently merely refers to the [http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/ equivalent IETF document]. In the IETF document use cases and requirements are listed. New use cases that have been proposed but not yet decided are also listed (with links).

There are however some items that should be discussed more within the context of current use cases and requirements (mostly taken from [http://www.ietf.org/mail-archive/web/rtcweb/current/msg01177.html this mail]):

* QoS. Unclear to a large extent at this time. Skip for first version?
* The use case (4.2.5) of using SIP between service providers must be sorted out to know what "SIP" really means and furthermore if all webrtc functionality would be available or if there are certain things you can't do. Also F24 should probably be updated, along with a new API requirement (text proposed from the mail "Web API MUST NOT prevent two webapps that happen to choose to peer with SIP from peering."). Also, the current A18 is vague and it can be discussed how much this is an API question
* What of more "advanced" audio media processing should remain in webrtc, and what should be moved to the Audio WG? Will have implications on F13 and F15 and A14.
* A more advanced IVR use case, or at least more detailed should perhaps replace the text in 4.3.2
* Codec requirements.....
* Section 7.2 (Security): should these requirements be moved into the F-req part?
* More work needed on data channels (unreliable, reliable, file-transfer) in terms of actual use and of requirements.

== Useful Resources ==

=== Mailing-lists ===
* [http://lists.w3.org/Archives/Public/public-webrtc/ Archives of the public-webrtc@w3.org mailing-list] are publicly available. This mailing-list should be used for all technical discussions within the group.
* [http://lists.w3.org/Archives/Member/member-webrtc/ Archives of the member-webrtc@w3.org mailing-list] are available to W3C Members only. Use of this second list should be restricted to administrative purposes (e.g. sending regrets or F2F logistics)

=== Tracker ===
* [http://www.w3.org/2011/04/webrtc/track/ Tracker] is used to track actions within the group. We're only using the ACTION section.
* [https://www.w3.org/Bugs/Public/describecomponents.cgi?product=WebRTC%20Working%20Group Buganizer] is available for tracking issues and bugs.

The [[Tracker usage guidelines]] (Work In Progress) gives details on how and when to use these tools.

=== IRC ===
The group uses IRC to take minutes and exchange info during calls and face-to-face meetings:

* IRC info: [irc://irc.w3.org:6665#webrtc '''#webrtc''' on irc.w3.org, port 6665]
* You may use the [http://irc.w3.org/ W3C Web client], specifying a username and channel #webrtc
* You may find more info on [http://www.w3.org/Project/IRC/ IRC usage at W3C]

=== Other Useful W3C Resources ===
* [http://www.w3.org/2005/10/Process-20051014/ World Wide Web Consortium Process Document] 14 October 2005
* [http://www.w3.org/Guide/ Art of Consensus Guide] (W3C member confidential)
* [http://www.w3.org/Project/IRC/ Notes on IRC at W3C]
* [http://www.w3.org/2001/12/zakim-irc-bot.html The Zakim IRC Teleconference Agent]
* [http://www.w3.org/Guide/1998/08/TeleconferenceHowTo.htm Setting up a telco using Zakim]
* [http://dev.w3.org/2002/scribe/scribedoc.htm#Quick_Start_Guide Quick Start Guide to taking minutes]

June 11 2012

Shkansso2 — Mon, 11 Jun 2012 07:12:58 GMT

Shkansso2: /* Hang-out */

== Logistics ==
Practical info about venue, hotels, ground transport etc: http://dl.dropbox.com/u/11888030/Practical%20Information%20for%20CLUE%20and%20WebRTC.htm

=== Meeting venue ===
The Interim meeting will be in the conference room Lappland at Nordic Forum. The street address is Torshamnsgatan 35 in Kista.

The walking directions are the following http://www.w3.org/2011/04/webrtc/wiki/images/d/d2/Nordic_Forum.jpg:

So when arriving at the subway ("T") station take the exit towards the rear of the train if arriving from downtown. From the station you come down the stairs and end up in the Galleria, turn left and go straight through the galleria. You exit the galleria onto the end of a square. In front of you are a street Kistagången" going uphill. At the far end of the street there is a high rise sticking up. Go towards it.

You will cross a bridge over a car street. In the next block after the bridge there is KTH/Electrum on the left. After that building you will have a walkway /bicycle path "Gröndlandsgången" to the left. Take that and then right after just 30-40 meters, up some stairs. Then you arrive on a parking lot, turn left and you get to the entrance. The meeting is in on the entrance floor.

If you get lost, please call +46 73 0949079 for help.

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

We plan to use WebEx and Google hang-out for remote participation.

==== Hang-out ====
http://plus.google.com/hangouts/_/google.com/webrtc

==== WebEx ====
Topic: WebRTC / RTCWeb
Date: Monday, June 11, 2012
Time: 9:00 am, Sweden Summer Time (Stockholm, GMT+02:00)
Meeting Number: 201 536 651
Meeting Password: hi

-------------------------------------------------------
To start the online meeting
-------------------------------------------------------
1. Go to https://cisco.webex.com/ciscosales/j.php?ED=195618382&UID=482186897&PW=NNThmYzFlNWZl&RT=MiMxMzA%3D
2. Log in to your account.
3. Click "Start Now".
4. Follow the instructions that appear on your screen.

----------------------------------------------------------------
ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
----------------------------------------------------------------

The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpitas area and (866) 349-3520 for the RTP area.

Please dial the local access number for your area from the list below:
- San Jose/Milpitas (408) area: 525-6800
- RTP (919) area: 392-3330

-------------------------------------------------------
To join the teleconference only
-------------------------------------------------------
1. Dial into Cisco WebEx (view all Global Access Numbers at
http://cisco.com/en/US/about/doing_business/conferencing/index.html
2. Follow the prompts to enter the Meeting Number (listed above) or Access Code followed by the # sign.

San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330

US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117

India: +91.80.4350.1111 Germany: +49.619.6773.9002

Japan: +81.3.5763.9394 China: +86.10.8515.5666

-------------------------------------------------------
For assistance
-------------------------------------------------------
1. Go to https://cisco.webex.com/ciscosales/mc
2. On the left navigation bar, click "Support".
To add this meeting to your calendar program (for example Microsoft Outlook), click this link:
https://cisco.webex.com/ciscosales/j.php?ED=195618382&UID=482186897&ICS=MS&LD=1&RD=2&ST=1&SHA2=DZvZpFa-P7S/TdCEzP155lteeNQDToeiVqsK7F3tqF8=

To check whether you have the appropriate players installed for UCF (Universal Communications Format) rich media files, go to https://cisco.webex.com/ciscosales/systemdiagnosis.php.

http://www.webex.com

== Agenda ==
(http://www.w3.org/2011/04/webrtc/wiki/images/3/38/AgendaJune11.pdf)

0900 Administrativia
* Welcome
* Actions

JSEPified PeerConn API (~3 h, whole morning) [[File:WebRTC_interim-june-2012_PeerConnection_API.pdf]]
* 0920: First session: API calls
** Walkthrough of calls
** Details that need work
* 1100: Second session: Other features needed
** error handling
** Constraints/Cap (network-stream specific things))
** other things

1330:
* Data channel API (1 h) [[File:WebRTC_interim-june-2012_Data_API.pdf]]
* Recording (1h)
** Relationship to MediaSource API
** Solicit a strawman?
1600:
* Stats API (1h) - presentation: [[File:June 11 Stats.pdf]]
* Media TF related stuff?
** Relationship to Audio API(s)
** Feedback / issues that need resolving in Media TF
* Requirements to go to CR (Dom should present) http://www.w3.org/2012/Talks/dhm-webrtc-testing/
** Tests (strategy, commitments)
** Implementations
1730 (- ~18.00):
* conclude, wrap up

== Expected attendants ==
=== Planning to turn up in person ===

==== Webrtc WG members ====
* Harald Alvestrand
* Stefan Håkansson
* Dominique Hazaël-Massieux
* Adam Bergkvist
* Vidhya Gholkar
* Jim Barnett
* Timothy B. Terriberry
* Jerome Marcon
* Richard P. Ejzak
* Dan Druta
* Markus Isomäki
* Dan Burnett
* Anant Narayanan
* Haakon Riiser

==== Observers ====
* Roni Even
* Mary Barnes
* Magnus Westerlund
* Ted Hardie
* Andrew Hutton
* Adamantini Martini
* Mauro Cabutto
* Gonzalo Camarillo
* Oscar Ohlsson
* Jonathan Lennox
* Charles Eckel
* Tommy Widenflycht
* Paul Kyzivat
* Göran Eriksson
* Stephan Wenger

=== Planning to attend from remote ===

* Dan Romascanu
* Cullen Jennings
* Randell Jesup
* Xavier Marjou
* Li Li
* Soo-Hyun Choi
* Yang Sun

June 11 2012

Shkansso2 — Mon, 11 Jun 2012 06:51:30 GMT

Shkansso2: /* Agenda */

== Logistics ==
Practical info about venue, hotels, ground transport etc: http://dl.dropbox.com/u/11888030/Practical%20Information%20for%20CLUE%20and%20WebRTC.htm

=== Meeting venue ===
The Interim meeting will be in the conference room Lappland at Nordic Forum. The street address is Torshamnsgatan 35 in Kista.

The walking directions are the following http://www.w3.org/2011/04/webrtc/wiki/images/d/d2/Nordic_Forum.jpg:

So when arriving at the subway ("T") station take the exit towards the rear of the train if arriving from downtown. From the station you come down the stairs and end up in the Galleria, turn left and go straight through the galleria. You exit the galleria onto the end of a square. In front of you are a street Kistagången" going uphill. At the far end of the street there is a high rise sticking up. Go towards it.

You will cross a bridge over a car street. In the next block after the bridge there is KTH/Electrum on the left. After that building you will have a walkway /bicycle path "Gröndlandsgången" to the left. Take that and then right after just 30-40 meters, up some stairs. Then you arrive on a parking lot, turn left and you get to the entrance. The meeting is in on the entrance floor.

If you get lost, please call +46 73 0949079 for help.

=== IRC ===
As usual, #webrtc at irc.w3.org

=== Remote participation ===

We plan to use WebEx and Google hang-out for remote participation.

==== Hang-out ====
https://plus.google.com/hangouts/_/ab7f2880394d4611328fba8ef9da5b9a5ffa2635?authuser=0&hl=sv#

Best guess right now: a link to the hangout session will be posted on the IRC when the meeting starts.

==== WebEx ====
Topic: WebRTC / RTCWeb
Date: Monday, June 11, 2012
Time: 9:00 am, Sweden Summer Time (Stockholm, GMT+02:00)
Meeting Number: 201 536 651
Meeting Password: hi

-------------------------------------------------------
To start the online meeting
-------------------------------------------------------
1. Go to https://cisco.webex.com/ciscosales/j.php?ED=195618382&UID=482186897&PW=NNThmYzFlNWZl&RT=MiMxMzA%3D
2. Log in to your account.
3. Click "Start Now".
4. Follow the instructions that appear on your screen.

----------------------------------------------------------------
ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
----------------------------------------------------------------

The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpitas area and (866) 349-3520 for the RTP area.

Please dial the local access number for your area from the list below:
- San Jose/Milpitas (408) area: 525-6800
- RTP (919) area: 392-3330

-------------------------------------------------------
To join the teleconference only
-------------------------------------------------------
1. Dial into Cisco WebEx (view all Global Access Numbers at
http://cisco.com/en/US/about/doing_business/conferencing/index.html
2. Follow the prompts to enter the Meeting Number (listed above) or Access Code followed by the # sign.

San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330

US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117

India: +91.80.4350.1111 Germany: +49.619.6773.9002

Japan: +81.3.5763.9394 China: +86.10.8515.5666

-------------------------------------------------------
For assistance
-------------------------------------------------------
1. Go to https://cisco.webex.com/ciscosales/mc
2. On the left navigation bar, click "Support".
To add this meeting to your calendar program (for example Microsoft Outlook), click this link:
https://cisco.webex.com/ciscosales/j.php?ED=195618382&UID=482186897&ICS=MS&LD=1&RD=2&ST=1&SHA2=DZvZpFa-P7S/TdCEzP155lteeNQDToeiVqsK7F3tqF8=

To check whether you have the appropriate players installed for UCF (Universal Communications Format) rich media files, go to https://cisco.webex.com/ciscosales/systemdiagnosis.php.

http://www.webex.com

== Agenda ==
(http://www.w3.org/2011/04/webrtc/wiki/images/3/38/AgendaJune11.pdf)

0900 Administrativia
* Welcome
* Actions

JSEPified PeerConn API (~3 h, whole morning) [[File:WebRTC_interim-june-2012_PeerConnection_API.pdf]]
* 0920: First session: API calls
** Walkthrough of calls
** Details that need work
* 1100: Second session: Other features needed
** error handling
** Constraints/Cap (network-stream specific things))
** other things

1330:
* Data channel API (1 h) [[File:WebRTC_interim-june-2012_Data_API.pdf]]
* Recording (1h)
** Relationship to MediaSource API
** Solicit a strawman?
1600:
* Stats API (1h) - presentation: [[File:June 11 Stats.pdf]]
* Media TF related stuff?
** Relationship to Audio API(s)
** Feedback / issues that need resolving in Media TF
* Requirements to go to CR (Dom should present) http://www.w3.org/2012/Talks/dhm-webrtc-testing/
** Tests (strategy, commitments)
** Implementations
1730 (- ~18.00):
* conclude, wrap up

== Expected attendants ==
=== Planning to turn up in person ===

==== Webrtc WG members ====
* Harald Alvestrand
* Stefan Håkansson
* Dominique Hazaël-Massieux
* Adam Bergkvist
* Vidhya Gholkar
* Jim Barnett
* Timothy B. Terriberry
* Jerome Marcon
* Richard P. Ejzak
* Dan Druta
* Markus Isomäki
* Dan Burnett
* Anant Narayanan
* Haakon Riiser

==== Observers ====
* Roni Even
* Mary Barnes
* Magnus Westerlund
* Ted Hardie
* Andrew Hutton
* Adamantini Martini
* Mauro Cabutto
* Gonzalo Camarillo
* Oscar Ohlsson
* Jonathan Lennox
* Charles Eckel
* Tommy Widenflycht
* Paul Kyzivat
* Göran Eriksson
* Stephan Wenger

=== Planning to attend from remote ===

* Dan Romascanu
* Cullen Jennings
* Randell Jesup
* Xavier Marjou
* Li Li
* Soo-Hyun Choi
* Yang Sun

File:AgendaJune11.pdf

Shkansso2 — Mon, 11 Jun 2012 06:50:46 GMT

Shkansso2: