WebRTC Discussions Summary from IETF 99 Meeting
31 July 2017 | Media, Security
There are disagreements about just how much work is left to do on the core WebRTC-related specs, but there is general agreement that mainly what needs to happen now is work on the specs rather than discussions of new topics.
Firefox 49 Improves WebRTC Standards Support and Bandwidth Control
29 August 2016 | Media
Better synchronization with WebRTC standards and Chrome and improved bandwidth control coming to you in Firefox 49.
Updates from Latest W3C Working Draft 03 August 2016
9 August 2016 | Media
4 items we selected from the change log of the new W3C WebRTC Working Draft to include in this update. See how it impacts your application.
WebRTC Virtual Meeting Decisions Summary
21 June 2016 | Media
WebRTC virtual meetings are held to speed up standardization work. Several decisions were made at the last virtual meeting.
Media Capture and Streams Specification Reaches W3C Candidate Recommendation Stage
20 May 2016 | Media
W3C Media Capture and Streams, the one defining getUserMedia and MediaStreamTracks reaches a stable stage
Support for Multiple RTX Codecs
26 April 2016 | Media
Chrome 50 changes WebRTC FEC implementation. Applications should act to ensure interoperability and benefit from higher quality
PeerConnection Error Callback now Mandatory
18 April 2016 | Media
WebRTC PeerConnection Error Callback for createOffer and createAnswer now mandatory in Chrome 50
Is It The End Of SDP in WebRTC As We Know It?
11 April 2016 | Media
SDP APIs in WebRTC might find their way out of the standard
WebRTC Trickle ICE End of Candidates Gathering Indication
21 March 2016 | Media
ICE is used in WebRTC and in SIP for finding the possible media routes for a session. This process takes time to complete and is one of the reasons for delay in establishing media connections in SIP and in WebRTC. Latest changes in WebRTC allow to shorten completion of the ICE procedure.
Debugging ICE in WebRTC
14 March 2016 | Media
As the WebRTC specification has evolved a gap developed between the ICE debug information available in the statistics draft and what is actually part of the ICE candidate attributes. There is now work for closing this gap.
WebRTC Standard Virtual F2F Meeting #1
16 February 2016 | Media
The team working on the WebRTC standards started to have virtual meetings in order to work on the open items and by that speed up standards work. Here are the highlights of the first meeting help in January.
Passive Aggressive Nomination Will Improve ICE Startup
26 January 2016 | Media
If a long connection time (time to open a call) bugs you, ICE is likely the one to blame for that. There have been different attempts to shorten the connection creation time. This post is just about that.
RTCP MUX and its Impact on Interoperability with Traditional VoIP
4 January 2016 | Media
WebRTC apps may now reject your non-RTCP MUX offer. Be prepared.
Simulcast and sending multiple video resolutions to the SFU
22 December 2015 | Media
Efficient SFU implementation with video simulcast
WebRTC standard work highlights from W3C Sapporo Japan and IETF 94 Yokohama Japan meetings
25 November 2015 | Application, Media
IETF & W3C Oct & Nov highlights.
WebRTC Early Media Now Possible
16 November 2015 | Media
As Transceivers were added to WebRTC now also early media is possible.
ICE Warm-up and Finding The Right m-line Using WebRTC Transceivers
8 November 2015 | Media
The most obvious change is that any application can now start everything up quickly by creating an RTCRtpTransceiver immediately after creating a Peer Connection
WebRTC Video Sharpness vs. Motion Prioritization
26 October 2015 | Media
More control over WebRTC video resolution and frame rate preferences
WebRTC Conferencing: Exchanging Audio Information between Client and Conference Mixer
20 October 2015 | Media
Knowing more about WebRTC conference mixed audio. New APIs for providing audio level and extracting mixed audio participants' information.
Media Capture from DOM Elements
13 October 2015 | Media
In an HTML video element there sometimes may not be a source connected to it, or the video might be paused. Idea behind change in standard is to better share the actual user experience of such cases.
WebRTC SDP Codec Priority Reordering and Subseting
5 October 2015 | Media
W3C decided to allow an API mechanism to get the list of negotiated codecs and reorder codec priority or even remove codecs from the list. Learn how this is done.
WebRTC Screen Sharing Discussion in W3C
29 September 2015 | Media
Standard work on screen capture is progressing nicely. Discussions recently have been primarily about screen capture user permissions and APIs for different capture scenarios and needs such as application, application windows, browser full screen and monitor.
W3C Interim Meeting Brings New Functionality and API Changes to WebRTC
17 September 2015 | Media
On September 9 to 10th there was a W3C interim meeting with 2 major areas on the surgery bed: • PeerConnection media negotiation • Local media
Changes to RTCRtpSender Impact WebRTC DTMF APIs
18 August 2015 | Media
canInsertDTMF attribute removed. Learn more about changes to DTMF related APIs.
Additions to RTCRtpSender Enhance Video Control
11 August 2015 | Media
Changes to RTCRtpSender Object gives more control over video parameters
updateIce was changed to setConfiguration, mind the incompatibility gap
3 August 2015 | Media
updateIce() was renamed to setConfiguration() and functionality enhanced.
addIceCandidate Is Now a Queued Method
14 July 2015 | Media
All methods that could take time to execute were made asynchronous, as is appropriate in a single-threaded language such as JavaScript. Some operations of WebRTC must be executed in a predefined order to ensure correct operation. SDP Offer/Answer is one such operation in WebRTC
SDP BUNDLE
17 June 2015 | Media
BUNDLE is an SDP feature used, among others, in WebRTC. The idea is simple; send all media flows (those m= lines in the SDP) using the same “5 tuple”, meaning from the same IP and port, to the same IP and port, and over the same transport protocol.
FEC for WebRTC
9 June 2015 | Media
WebRTC often uses Forward Error Correction (FEC) in its encoding of media. FEC is used for protecting media packets from packet corruption by adding redundant information that is later used on the receiving side to compensate for data that was lost and improve voice and video quality. FEC also introduces an overhead, because the redundant information FEC adds increases the bandwidth used. The tricky part is to balance usage of FEC.
Chrome Experiments to Determine Realistic STUN Check Intervals
2 June 2015 | Media
The time it takes to know a WebRTC media connection broke down has a tradeoff associated to it
SDP Changes and Their Impact on WebRTC
18 May 2015 | Media
SDP is going through some changes. Read this post to learn more and how it will impact your application.
RETURN Helps Avoiding WebRTC Call Failure
12 May 2015 | Media
If your WebRTC calls fail it might be due to a border TURN server in the enterprise. RETURN comes to the rescue. Learn how.