WebRTC Standards & Implementation Q&A - The Internals of WebRTC Browsers Implementation
4 likes804 views
This document summarizes a Q&A session on WebRTC standards and implementation with Amir Zmora and Dan Burnett. Some of the key topics discussed include: - Differences between how Chrome and other browsers implement WebRTC compared to the libwebrtc C++ library, especially around capabilities like screen sharing. - The layered architecture of how WebRTC is implemented in different browsers, with Chrome having additional layers for security and capabilities compared to the libwebrtc implementation. - The classes and namespaces used to design the libwebrtc C++ library and how concepts like PeerConnection, MediaStreams, and DataChannels are represented. - Long term plans from Google to restructure the Web
![Call API today (Content courtesy Google)
• everything that's not ICE, DTLS or SDP.
• Underneath Call, there are already "RtpSender”, "RtpReceiver”,
"RtpTransport”, but they're buried down many levels.
• So the RtpSender/RtpReceiver objects […] are actually just thin proxies
that rely on objects several layers below to do the real work.
• Any time [Google] adds a new API (such as GetContributingSources),
[they] need to plumb it through all those layers, which is time-consuming,
difficult to maintain, prone to bugs, etc.
• So what [they] plan to do is pull the RTP objects out of Call, and merge
them with the API-level objects.
Call
& related
interfaces
• AudioRtpSender ("API level")
• VoiceChannel
(BaseChannel subclass)
• WebRtcVoiceMediaChannel
(MediaChannel subclass)
• Call
• AudioSendStream
• voe::ChannelProxy
• voe::Channel
• ModuleRtpRtcpImpl
• RTPSender ("Call level")](https://image.slidesharecdn.com/webrtcstandardslivesession18-170829061335/85/WebRTC-Standards-Implementation-Q-A-The-Internals-of-WebRTC-Browsers-Implementation-27-320.jpg)
WebRTC Standards & Implementation Q&A - The Internals of WebRTC Browsers Implementation
- 1. WebRTC Standards Q&A Amir Zmora TheNewDialTone Dan Burnett StandardsPlay
- 2. Guest Speaker • Previously part of the WebRTC Standard team • Left to handle CoSMo growth (1~12 in 6months) • Consulting on WebRTC, with focus on: • C++ libwebrtc and Native Apps • Build /Test / Package • Codecs • (Double) Encryption • Qt wrappers • Media Servers (Meedoze, Janus, Jitsi) • System level design and implementation Dr. Alex Gouaillard WebRTC by Dr Alex CoSMo Software
- 3. Watch video recording of this session http://ccst.io/e/webrtcstandards18
- 4. Session sponsored by WebRTC.ventures is a custom design and development shop dedicated to building WebRTC based applications for web and mobile. We have built end-‐to-‐end broadcast solutions for events and entertainment clients, telehealth solutions for multiple clients, live support tools, as well as communication tools for a variety of other applications. WebRTC.ventures is a recognized development partner of TokBox and has also built native WebRTC solutions
- 5. We use CrowdCast….It’s WebRTC
- 6. About Us • Amir Zmora • Dan Burnett
- 10. Chrome Variation • Wait .. What? Chrome does not use all of libwebrtc? Why?
- 11. Chrome Layers
- 12. Chrome Layers
- 13. Chrome Variation • Wait .. What? Chrome does not use libwebrtc? Why? • Chrome layering • Sandbox and per tab security • Echo canceling • Special capturers (screen/window/tab) • Capturer access security • Pretty much the same applies to all browsers
- 14. Chrome Variations • OK, Chrome does not use libwebrtc? Why should I Care? • If you use the webrtc lib in your project, and expect the same results as chrome, be ready for big surprises! • No Screen/windows/tab sharing • Obsolete mac capturer • No HTTP PROXY support • …
- 16. Firefox Variation – Desktop 1process
- 17. Firefox Variation – B2G
- 18. FireFox Arch • PeerConnection • PeerConnection.js -- shim translation layer to let us do API adaptation to the C++ • PeerConnectionImpl -- C++ implementation of the PeerConnection interface. • SIPCC -- handles SDP and media negotiation. Provided by Cisco but not a downstream. • Media • Webrtc.org/GIPS -- handles media encoding and decoding. Downstream from Google. • MediaConduit -- Generic wrapper around Webrtc.org • MediaPipeline -- Wrapper to hold the MediaConduit, mtransport subsystem, and the SRTP contexts, as well as interface with MediaStreams. • Transport • mtransport -- generic transport subsystem with implementations for ICE, DTLS, etc. • NSS -- new DTLS stack. Mentioned because we need to land the new version of NSS • nICEr -- ICE stack;; downstream from reSIProcate project • nrappkit --portable runtime, utility library;; downstream from nrappkit.sourceforge.net • DataChannel • DataChannel implementation in the DOM • libsctp -- SCTP implementation;; downstream from the BSD SCTP guys
- 19. Webkit Arch
- 20. Webkit Arch
- 21. Webkit Webrtc Class Arch
- 23. LibWEBRTC Name Space Design
- 26. PeerConnection and friends (Slide courtesy Google) PeerConnection WebRtcSession TransportController BaseChannel MediaChannel DtlsTransport P2PTransportChannel JsepTransport ChannelManager Call & related interfaces
- 27. Call API today (Content courtesy Google) • everything that's not ICE, DTLS or SDP. • Underneath Call, there are already "RtpSender”, "RtpReceiver”, "RtpTransport”, but they're buried down many levels. • So the RtpSender/RtpReceiver objects […] are actually just thin proxies that rely on objects several layers below to do the real work. • Any time [Google] adds a new API (such as GetContributingSources), [they] need to plumb it through all those layers, which is time-consuming, difficult to maintain, prone to bugs, etc. • So what [they] plan to do is pull the RTP objects out of Call, and merge them with the API-level objects. Call & related interfaces • AudioRtpSender ("API level") • VoiceChannel (BaseChannel subclass) • WebRtcVoiceMediaChannel (MediaChannel subclass) • Call • AudioSendStream • voe::ChannelProxy • voe::Channel • ModuleRtpRtcpImpl • RTPSender ("Call level")
- 28. Long term plans (With Unified Plan) (Slide Courtesy Google) PeerConnection TransportController (?) DtlsTransport P2PTransportChannel JsepTransport RtpTransport RtpTransceiver RtpSender RtpReceiver
- 29. ?
- 31. WebRTC Standards & Implementation Q&A Amir Zmora TheNewDialTone Dan Burnett StandardsPlay