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WebRTC Standards Q&A
Amir Zmora
TheNewDialTone
Dan Burnett
StandardsPlay
Recording is now available
Watch it now: http://ccst.io/e/webrtcstandards1
We use CrowdCast….It’s WebRTC
About Us
• Amir Zmora • Dan Burnett
WebRTCStandards.info
WebRTC Browser Support
IsWebRTCReadyYet.com
thenewdialtone.com/webrtc-in-safari
Codec Support
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…
WebRTC Broadcast
NPM WebRTC Scalable Broadcast
WebRTC Broadcast
Flashphoner
Track-level controls
Senders, Receivers, & Transceivers
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
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
ORTC -> WebRTC
• So, ORTC APIs will eventually end up in WebRTC
• Guess what? Some controls moving now (into WebRTC 1.0)!
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()
New track-level controls
• RTCRtpSenders
• RTCRtpReceivers
pc.addTrack(myVideoTrack)
.then(function(sender) {myVideoSender = sender;};
pc.ontrackadded = function (e) {
myAudioReceiver = e.receiver;};
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
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);
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;
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);
});
Thank You
Amir Zmora
TheNewDialTone
Dan Burnett
StandardsPlay

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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
  • 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
  • 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); });
  • 21.