B.1 Changes since 01 March 2016
- Added the
gather()method, as noted in: Issue 165
- Removed “public” from
, as noted in: Issue 224
- Removed the minQuality attribute, as noted in: Issue 351
receive()asynchronous, as noted in: Issue 399, Issue 463, Issue 468 and Issue 469
- Provided additional information on ICE candidate errors, as noted in: Issue 402
- Added state attribute to
, as noted in: Issue 403
- Provided an example of RTX/RED/FEC configuration, as noted in: Issue 404
payloadTypeuniqueness, as noted in: Issue 405
- Updated the list of header extensions, as noted in: Issue 409
- Added “goog-remb” to the list of feedback mechanisms, as noted in: Issue 410
- Added kind argument to the
constructor, as noted in: Issue 411
send()restrictions on kind, as noted in: Issue 414
getAlgorithm()method, as noted in: Issue 427
protocol and label to USVString, as noted in: Issue 429
- Clarified nullable attributes and methods returning empty lists, as noted in: Issue 433
- Clarified support for the “direction” parameter, as noted in: Issue 442
- Clarified the apt capability of the “red” codec, as noted in: Issue 444
- Clarified usage of
attributes, as noted in: Issue 445
- Clarified firing of
onssrcconflictevent, as noted in: Issue 448
- Clarified that CNAME is only set on an
, as noted in: Issue 450
- Updated references, as noted in: Issue 457
- Described behavior of
, as noted in: Issue 461
- Corrected dictionary initialization in the examples, noted in: Issue 464 and Issue 465
- Corrected use of enums in the examples, noted in: Issue 466
- Clarified handling of identity constraints, as noted in: Issue 467 and Issue 468
- Clarified use of
RTCRtpEncodingParameters, as noted in: Issue 470
- Changed hostCandidate type, as noted in: Issue 474
- Renamed state change event handlers to onstatechange, as noted in: Issue 475
- Updated description of
closed state, as noted in: Issue 476
- Updated description of
object, as noted in: Issue 477
- Updated description of relatedPort, as noted in: Issue 484
- Updated description of
, as noted in: Issue 485
- Clarified exceptions in
construction, as noted in: Issue 492
- Provided a reference to
error.message, as noted in: Issue 495
description, as noted in: Issue 496
- Clarified default for clockRate attribute, as noted in: Issue 500
- Removed use of “null if unset”, as noted in: Issue 503
constructor, as noted in: Issue 504
- Clarified behavior of
getCapabilities(), as noted in: Issue 509
- Addressed issues with
, as noted in: Issue 519
Last Thursday we had the first virtual w3c webrtc wg interim meeting. Once we sorted out a few technical details it went quite well!
Meeting Home Page:
ORTC, WebRTC, H.264, VP8, RID, RtpEncoding, Simulcast and much more. Google, Microsoft and Hookflash leading the discussion, join us!
We have an immediate WebRTC development contract opportunity that has just come up in the Seattle area. The contract requires 4-5 full-time developers onsite, remote will not fit the bill on this one.
For this contract we are looking for a team lead, 2 x Node.js, 2 x common JS developers
You have built commercial web applications using WebRTC libraries and are intimately familiar with the WebRTC and ORTC specs and respective libraries.
Start date: ASAP
If you are interested please forward your resume email@example.com
Our initial ORTC implementation includes the following components:
- ORTC API Support. Our primary focus right now is audio/video communications. We have implemented the following objects: IceGatherer, IceTransport, DtlsTransport, RtpSender, RtpReceiver, as well as the RTCStatsinterfaces that are not shown directly in the diagram.
- RTP/RTCP multiplexing is supported and is required for use with DtlsTransport. A/V multiplexing is also supported.
- STUN/TURN/ICE support. We support STUN (RFC 5389), TURN (RFC 5766) as well as ICE (RFC 5245). Within ICE, regular nomination is supported, with aggressive nomination partially supported (as a receiver). DTLS-SRTP (RFC 5764) is supported, based on DTLS 1.0 (RFC 4347).
- Codec support. For audio codecs, we support G.711, G.722, Opus and SILK. We also support Comfort Noise (CN) and DTMF according to the RTCWEB audio requirements. For video we currently support the H.264UC codec used by Skype services, supporting advanced features such as simulcast, scalable video coding and forward error correction. We’re working toward to enabling interoperable video with H.264.
W3C WebRTC working group chairs [Harald Alvestrand (Google), Stefan Håkansson (Ericsson), Erik Lagerway (Hookflash)], made a decision recently to add a new editor to the working group, as Peter St. Andre (&yet) has resigned as editor.
Bernard Aboba (Microsoft) has now been appointed as editor.
Bernard’s attention to detail and advocacy for transparency, fairness and community has been refreshing. It has been my pleasure (as chair of the W3C ORTC CG) to work with Bernard whom also is an author in the W3C ORTC CG alongside Justin Uberti and Robin Raymond (editor). I look forward to working more with him in the WG.