WebRTC Standards Q&A Live Session 1 by Dan Burnett & Amir Zmora
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The document discusses the latest advancements and standards in WebRTC, highlighting the collaboration between WebRTC and ORTC, aiming for full compatibility in future versions. It outlines new track-level controls for developers, enabling more control over media transport, and introduces features like RTCRtpTransceivers for enhanced management of audio and video tracks. The ongoing evolution of WebRTC is positioned to improve call setup times and media handling efficiency.
![Controls on senders/receivers
• Post-O/A codec control
• Simulcast
• degradationPreference
params = sender.getParameters();
var save0 = params.codecs[0];
params.codecs[0] = params.codecs[1];
params.codecs[1] = save0;
sender.setParameters(params);
params = sender.getParameters();
params.encodings.scaleResolutionDownBy = 2.0;
params.encodings.scaleResolutionDownBy = 4.0;
sender.setParameters(params);
params = sender.getParameters();
params.degradationPreference = "maintain-framerate";
sender.setParameters(params);](https://image.slidesharecdn.com/webrtcstandardslivesession1-160124202133/85/WebRTC-Standards-Q-A-Live-Session-1-by-Dan-Burnett-Amir-Zmora-18-320.jpg)
![Controls on transceivers
• Pre-O/A codec control
• Pre-allocation to speed up call setup
transceiver.setCodecPreferences(["audio/OPUS", "audio/PCMU"]);
audioT = pc.addTransceiver("audio");
// do O/A now, which lets DTLS get started
navigator.mediaDevices.getUserMedia("audio":true)
.then(function(realAudioTrack) {
audioT.sender.replaceTrack(realAudioTrack);
});](https://image.slidesharecdn.com/webrtcstandardslivesession1-160124202133/85/WebRTC-Standards-Q-A-Live-Session-1-by-Dan-Burnett-Amir-Zmora-20-320.jpg)
WebRTC Standards Q&A Live Session 1 by Dan Burnett & Amir Zmora
- 1. WebRTC Standards Q&A Amir Zmora TheNewDialTone Dan Burnett StandardsPlay
- 2. Recording is now available Watch it now: http://ccst.io/e/webrtcstandards1
- 3. We use CrowdCast….It’s WebRTC
- 4. About Us • Amir Zmora • Dan Burnett
- 8. WebRTC Broadcast Hangouts On Air CrowdCast Yodlr Ubivent BigMarker ClassOnLive Citrix (Future) Periscope & Meerkat Service TokBox (Spotlight) PaaS WebRTC Experiments WebRTC Scalable Broadcast (NPM) Flashphoner Open Source/Demos/SDKs Also look at Peer Assisted CDN: Peer5, StreamRoot, Peeracle…
- 9. WebRTC Broadcast NPM WebRTC Scalable Broadcast
- 11. Track-level controls Senders, Receivers, & Transceivers
- 12. Object RTC (ORTC) • Started by Hookflash in 2013 • Originally positioned as alternative to WebRTC 1.0 • Developed in a non-standards-track Community Group at W3C • Active participants: Microsoft and Google, others • Peer Connection establishment • WebRTC 1.0: SDP Offer/Answer protocol (JSEP) • ORTC: Capabilities API + proprietary agreement • Now, ORTC aiming to be compatible with WebRTC
- 13. WebRTC & ORTC • Great news! • For WebRTC > 1.0, ORTC and WebRTC participants have agreed • All official WebRTC standardization work will happen in WebRTC Working Group • New API controls will not be required to have an SDP implementation • WebRTC 1.0 will be included as a required part of the next version • Erik Lagerway of Hookflash now a co-chair of WebRTC working group • Bernard Aboba of Microsoft now a co-editor of WebRTC spec
- 14. ORTC -> WebRTC • So, ORTC APIs will eventually end up in WebRTC • Guess what? Some controls moving now (into WebRTC 1.0)!
- 15. Why new controls? • WebRTC designed to ease set up of basic calls • … but developers want more control • Specifically, control over how media is transported over a peer connection • things like codecs, priority, congestion-caused changes • and absolutely anything that can accelerate call setup time! • This has caused fundamental changes in the architecture • Focus now on tracks, not streams • addTrack(), removeTrack()
- 16. New track-level controls • RTCRtpSenders • RTCRtpReceivers pc.addTrack(myVideoTrack) .then(function(sender) {myVideoSender = sender;}; pc.ontrackadded = function (e) { myAudioReceiver = e.receiver;};
- 17. Info on senders/receivers • Transports • RTCDtlsTransport • sender.transport and sender.rtcpTransport, can be shared according to BUNDLE and RTCP-MUX • Contains • (ICE) transport, state, getRemoteCertificates() • ICE transport info: component, role, state, gatheringState, local/remote candidates/parameters, and selected candidate pair • state is new, connecting, connected, closed, failed • certificates are DTLS certificate chain used by the remote side • Parameters • (simulcast) encodings, codecs, rtcp parameters, RTP header extensions
- 18. Controls on senders/receivers • Post-O/A codec control • Simulcast • degradationPreference params = sender.getParameters(); var save0 = params.codecs[0]; params.codecs[0] = params.codecs[1]; params.codecs[1] = save0; sender.setParameters(params); params = sender.getParameters(); params.encodings.scaleResolutionDownBy = 2.0; params.encodings.scaleResolutionDownBy = 4.0; sender.setParameters(params); params = sender.getParameters(); params.degradationPreference = "maintain-framerate"; sender.setParameters(params);
- 19. Transceivers • Why? • JSEP automatically assigns tracks to SDP media lines • … sometimes in a way you don't want • RTCRtpTransceivers provide explicit control over which SDP media sections used for which tracks • … and show you which sender and receiver are tied together in SDP • What? t1 = pc.addTransceiver("audio", {send: false, receive: true}); t2 = pc.addTransceiver(myVideoTrack); (… later) receivedAudio = t1.receiver.track; receivedVideo = t2.receiver.track;
- 20. Controls on transceivers • Pre-O/A codec control • Pre-allocation to speed up call setup transceiver.setCodecPreferences(["audio/OPUS", "audio/PCMU"]); audioT = pc.addTransceiver("audio"); // do O/A now, which lets DTLS get started navigator.mediaDevices.getUserMedia("audio":true) .then(function(realAudioTrack) { audioT.sender.replaceTrack(realAudioTrack); });
- 22. Thank You Amir Zmora TheNewDialTone Dan Burnett StandardsPlay