public-webrtc@w3.org from October 2014 by subject

[Bug 15861] API for JS interaction with congestion control

[Bug 17287] PeerConnectionErrorCallback argument

[Bug 17704] suggest a parameter in getUserMedia

[Bug 19580] Callbacks need to be called asynchronously

[Bug 19593] create reference for mediaconstraints in createOffer() method

[Bug 19729] missing a reference for XMLHttpRequest in 4.1 Introduction

[Bug 19730] suggest to remove issue 1 from 4.3.1

[Bug 19731] wrong transition description between "checking" and "connected" in 4.4.3

[Bug 20806] Section 15 (Security Considerations) is empty

[Bug 20807] End-to-end stream configuration unspecified

[Bug 20808] "Trickle ICE" unspecified

[Bug 20809] Stream rejection not possible

[Bug 20810] SDP inadequately defined

[Bug 20811] RTP usage not defined

[Bug 20812] Media protocol not specified

[Bug 20813] Protocol security not defined

[Bug 20815] Stream multiplexing

[Bug 20816] "Hold" unspecified

[Bug 20817] Media loopback testing

[Bug 20819] no priority API

[Bug 20820] congestion control of data and audio/video

[Bug 21086] getLocalStreams and getRemoteStreams should return empty sequence after Peerconnection::close

[Bug 21877] API is unable to handle inbound streams prior to arrival of answer

[Bug 21878] Unable to learn of unknown inbound media

[Bug 21879] Unable to access certificate information in the API

[Bug 21880] Certificate management is underspecified

[Bug 21950] Add success/error callbacks to addIceCandidate (and possibly other API calls)

[Bug 22347] RTCIceServer should have multiple URLs

[Bug 22441] Bug in section 8.1.2 Requesting Assertions

[Bug 22442] Bug in section 8.1.3 Verifying Assertions

[Bug 23572] Documented format with which to specify ICE servers does not match implementation and contains typos

[Bug 23832] Requiring that negotiated channels be created on the receiver before any data can be received is problematic for some use cases

[Bug 23919] DataChannel.onerror callback needs an error argument specified.

[Bug 23920] TURN authentication failures should be surfaced as some event

[Bug 24061] Need to update TURN / STUN URI references

[Bug 25102] RTCDataChannel::send() steps are not proper.

[Bug 25440] MediaStreamTrack.readyState has no muted attribute

[Bug 25497] RTCRtpSender / Receiver objects need to be added to the specification

[Bug 25513] WebRTC spec should explicitly specify all causes of a PeerConnection-sourced track being muted

[Bug 25531] Validation for requestIdentity attribute is missing.

[Bug 25533] WebRTC spec should explicitly specify the state transition for cancelled offers.

[Bug 25544] Options attribute of createOffer / createAnswer should be validated before processing.

[Bug 25545] Initialization of of RTCConfiguration while invoking RTCPeerConnection.getConfiguration should be updated.

[Bug 25576] steps for createDTMFSender() are missing.

[Bug 25579] State transitions are missing in RTCPeerConnections state transition diagram.

[Bug 25596] updateIce should be called setConfiguration

[Bug 25806] ice pool size

[Bug 25808] add new acces for the active remote/local SDP

[Bug 25811] Change extensible enum to dom strings

[Bug 25828] Need to add pc.canTrickle)

[Bug 25833] change the definition of "enqueue a task" as EKR slides May 20

[Bug 25834] close is synchronous & idempotent

[Bug 25835] when closing, all outstanding actions are cancelled and their callbacks are fired with a "cancelled" error

[Bug 25836] add note about addtrack being async

[Bug 25856] Add way to find out if a MST is isolated or becomes isolated

[Bug 25859] Streams that become isolated generate errors on PC

[Bug 25893] Offer Answer options should supported sendOnly and inactive media states.

[Bug 25957] PeerConnection should have an onerror event handler

[Bug 25975] When can the value of DTMFSender.canInsertDTMF change?

[Bug 26027] addIceCandidate should not be callable when PeerConnection is closed

[Bug 26279] Options attribute is required for createAnswer

[Bug 26364] Add "rollback" to RTCSdpType

[Bug 26620] getStats should be allowed on a closed PeerConnection.

[Bug 26644] Candidate event attributes

[Bug 26644] MID in Candidate event attributes

[Bug 27211] Add BundlePolicy to RTCConfiguration

[Bug 27211] New: Add BundlePolicy to RTCConfiguration

[Bug 27213] DTMFSender should hang off RTCRTPSender, not MediaStreamTrack

[Bug 27213] New: DTMFSender should hang off RTCRTPSender, not MediaStreamTrack

[Bug 27214] ICE gathering state change should surface an event

[Bug 27214] New: ICE gathering state change should surface an event

[xrblock] I-D Action: draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00.txt

Agenda, updated

Agenda, updated (Media Capture)

Agenda, updated (WebRTC for real this time)

Call for Exclusions: Identifiers for WebRTC's Statistics API

Call for Exclusions: Media Capture Depth Stream Extensions

CfC ended: publish FPWD of Identifiers for WebRTC's Statistics API; respond by 10 Oct 2014.

CfC: publish FPWD of Identifiers for WebRTC's Statistics API; respond by 10 Oct 2014.

Change SDP role during renegotiation

FPWD of "Identifiers for WebRTC's Statistics API" published (was Fwd: Re: Publication request: FPWD of "Identifiers for WebRTC's Statistics API")

I would like to join

iceTransports=relay - use vs gather

keying

Keys stuff w/ indexeddb

Migrating WebRTC 1.0 Editor's draft to github

Proposal for WebIDL for bundle policy from JSEP

RTCRTPSender / Receiver and AddTrack have arrived! (pull request)

RTPSender and Receiver integrated in document

Stats document - status and intent to call for FPWD

Time split for TPAC

TPAC 2014 Agenda

TPAC agenda outline proposal

Using bugzilla (was Re: [Bug 20819] no priority API)

Video freeze issue...

WebRTC agenda TPAC

webrtc-ACTION-114: Make pull request on rtpsender.mid

webrtc-ACTION-115: Give an example of indexed-db key retrieval

webrtc-ACTION-116: Draft pull request on icecandidate error

webrtc-ACTION-117: Move getstats and associated idl to stats doc

webrtc-ACTION-118: Review matches between capabilities and stats

webrtc-ACTION-119: Look at our bug tracking strategy for webrtc with harald

Which stream/transport does PC.oniceconnectionstatechange points to?

Last message date: Friday, 31 October 2014 20:59:35 UTC