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Browsers and stack
implementation differences
Dr. Alex Gouaillard
Plan
• global architectures of browsers and stacks
• -- chrome / firefox / webkit
• -- webrtc stand alone, webrtc in chrome,
webrtc in firefox
• - how to trace the different browsers without
recompiling
WebRTC Arch
LibWEBRTC Variation
Chrome Variation
Chrome Variation
• Wait .. What? Chrome does not use libwebrtc?
Why?
Chrome Layers
Chrome Layers
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
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
LibWEBRTC src Files layout
Libwebrtc src file layout
• base/
• common_audio/
• common_video/
• libjingle/
• modules/
• p2p/
• sound/
• system_wrappers/
• tools/
• video/
• video_engine/
• voice_engine/
• Too many to list
• Basic signal processing (FFT, window, transforms, …)
• Frame buffer, I420, libyuv, …
• Session management
• See next slide
• ICE Transports: TURN, STUN, DTLS, ….
• Platform specific sound system wrappers
• Platform specific low level wrappers (threads, timer, …)
• Poach and weaponize!!
• Video encoder/decoder/rec and send stream
• Capturer, Channel, Encoder, Sync, REMB, Send
• Channel, DTMF, …
Libwebrtc src file layout
• audio_coding/
• audio_conference_mixer/
• audio_device/
• audio_processing/
• bitrate_controller/
• desktop_capture/
• media_file/
• pacing/
• remote_bitrate_estimator/
• rtp_rtcp/
• utility/
• video_capture/
• video_coding/
• video_processing/
• video_render/
• ACM2, NACK, NETQ, DTMF tone gen., ….
• mixer
• Platform specific
• Noise, typing, gain detection, NS, Echo Cancellation, …
• Bandwidth estimation and bitrate controller
• Desktop capture (abstract only)
• Audio files reading/writing
• Paced sender
• Obvious
• RTP, RTCP, Payloads, FEC, …
• More Audio files utilities, rtp_dump, recorder, player
• Generic and platform dependent capturers (OLD)
• Codecs, encoders and decoders, send and rec, jitter, …
• Video processing and image enhancement
• Generic and platform dependent renderers (OLD)
Libwebrtc src file layout
• Why do I care?
– 1. you might need to go in the code
– 2. you need to know the layout for debugging:
chrome.exe --enable-logging --
vmodule=*/webrtc/*=2,*/libjingle/*=2,*=-2 --no-
sandbox
LibWEBRTC Class Arch
LibWEBRTC Arch
LibWEBRTC DEPS
• binutils
• Boringssl
• Colorama
• Drmemory
• Expat
• Icu
• instrumented_libraries
• Jsoncpp
• Libjpeg
• libjpeg_turbo
• Libsrtp
• libudev
• libvpx
• Libyuv
• llvm-build
• Nss
• Ocmock
• openmax_dl
• Opus
• Protobuf
• Sqlite
• Syzygy
• Usrsctp
• Yasm
• zlib
Webrtc.org
Chrome Specific Webrtc Bits
• //src/content/renderer/media/
• //src/third_party/WebKit/public/platform/
• //src/third_party/WebKit/Source/modules/mediastream
• //src/content/browser/media/
• //src/content/browser/media/capture/
• //src/media/capture/video/
• //src/content/browser/renderer_host/media/
• .
• .
• .
• .
• .
• .
• .
Chrome Specific Webrtc Bits
// MediaStreamManager is used to open/enumerate media capture devices (video
// supported now). Call flow:
// 1. GenerateStream is called when a render process wants to use a capture
// device.
// 2. MediaStreamManager will ask MediaStreamUIController for permission to
// use devices and for which device to use.
// 3. MediaStreamManager will request the corresponding media device manager(s)
// to enumerate available devices. The result will be given to
// MediaStreamUIController.
// 4. MediaStreamUIController will, by posting the request to UI, let the
// users to select which devices to use and send callback to
// MediaStreamManager with the result.
// 5. MediaStreamManager will call the proper media device manager to open the
// device and let the MediaStreamRequester know it has been done.
Chrome GUM example
capturers / frames / Encoding / ….
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)GUM JS API
(tab/sandbox)
1. Send request
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)GUM JS API
(tab/sandbox)
2. Check if MST is already available
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)GUM JS API
(tab/sandbox)
Security Manager
(source, origin)
3. Check rights
2. Check if MST is already available - NO
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)
Audio
Capturer
Video
Capturer
Security Manager
(source, origin)
GUM JS API
(tab/sandbox)
4. Ask Corresponding capturer
type to start capturing
3. Check rights - OK
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)
Audio
Capturer
Video
Capturer
Security Manager
(source, origin)
A
GUM JS API
(tab/sandbox)
4. Ask Corresponding capturer
type to create one - OK
V
5. Store the MST
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)
Audio
Capturer
Video
Capturer
Security Manager
(source, origin)
A
GUM JS API
(tab/sandbox)
V
6. Trigger callback
Keep feeding frames
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)
Audio
Capturer
Video
Capturer
Security Manager
(source, origin)
A
GUM JS API
(tab/sandbox)
V
NOTE 1: second call for same device with same constraints will directly return the
MST, that allows to share streams across tabs without blocking
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)
Audio
Capturer
Video
Capturer
Security Manager
(source, origin)
A
GUM JS API
(tab/sandbox)
V
NOTE 2: Recently, a second call for the same device but with different constraints
(think simulcast) will indeed return a different resolution. Before it would return the
first resolution asked.
Chrome Specific Webrtc Bits
Media Stream Manager
(singleton@browser)
Audio
Capturer
Video
Capturer
Security Manager
(source, origin)
A
GUM JS API
(tab/sandbox)
V
NOTE 3: Not only this allow to share cams across processes, it allows for global echo
cancellation (yes, including the key strokes). Before tabs could cross feed.
Chrome Screensharing 2 steps (1)
Media Stream Manager
(singleton@browser)
Screen/Windows/Tab
Capturer
Security Manager
(source, origin)
Screensharing
(extension)
1
2
3
4
S
5
Chrome Screensharing 2 steps (2)
Media Stream Manager
(singleton@browser)
Screen/Windows/Tab
Capturer
S
GUM JS API
(tab/sandbox)
With ID
Firefox Variation
Firefox Variation – Desktop 1process
Firefox Variation – B2G
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
Webkit (safari ?)
Current work:
- Implement datachannel for webkit (yin liu)
- Implement webrtc.org as a back end for
webkit (adam tiamou)
- Implement windows support for OWR
Webkit Arch
Webkit Arch
Webkit Webrtc Arch
Webkit Arch – Data Channel
Webkit Arch – Data Channel

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WebRTC Browsers n Stacks Implementation differences

  • 1. Browsers and stack implementation differences Dr. Alex Gouaillard
  • 2. Plan • global architectures of browsers and stacks • -- chrome / firefox / webkit • -- webrtc stand alone, webrtc in chrome, webrtc in firefox • - how to trace the different browsers without recompiling
  • 6. Chrome Variation • Wait .. What? Chrome does not use libwebrtc? Why?
  • 9. 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
  • 10. 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
  • 12. Libwebrtc src file layout • base/ • common_audio/ • common_video/ • libjingle/ • modules/ • p2p/ • sound/ • system_wrappers/ • tools/ • video/ • video_engine/ • voice_engine/ • Too many to list • Basic signal processing (FFT, window, transforms, …) • Frame buffer, I420, libyuv, … • Session management • See next slide • ICE Transports: TURN, STUN, DTLS, …. • Platform specific sound system wrappers • Platform specific low level wrappers (threads, timer, …) • Poach and weaponize!! • Video encoder/decoder/rec and send stream • Capturer, Channel, Encoder, Sync, REMB, Send • Channel, DTMF, …
  • 13. Libwebrtc src file layout • audio_coding/ • audio_conference_mixer/ • audio_device/ • audio_processing/ • bitrate_controller/ • desktop_capture/ • media_file/ • pacing/ • remote_bitrate_estimator/ • rtp_rtcp/ • utility/ • video_capture/ • video_coding/ • video_processing/ • video_render/ • ACM2, NACK, NETQ, DTMF tone gen., …. • mixer • Platform specific • Noise, typing, gain detection, NS, Echo Cancellation, … • Bandwidth estimation and bitrate controller • Desktop capture (abstract only) • Audio files reading/writing • Paced sender • Obvious • RTP, RTCP, Payloads, FEC, … • More Audio files utilities, rtp_dump, recorder, player • Generic and platform dependent capturers (OLD) • Codecs, encoders and decoders, send and rec, jitter, … • Video processing and image enhancement • Generic and platform dependent renderers (OLD)
  • 14. Libwebrtc src file layout • Why do I care? – 1. you might need to go in the code – 2. you need to know the layout for debugging: chrome.exe --enable-logging -- vmodule=*/webrtc/*=2,*/libjingle/*=2,*=-2 --no- sandbox
  • 17. LibWEBRTC DEPS • binutils • Boringssl • Colorama • Drmemory • Expat • Icu • instrumented_libraries • Jsoncpp • Libjpeg • libjpeg_turbo • Libsrtp • libudev • libvpx • Libyuv • llvm-build • Nss • Ocmock • openmax_dl • Opus • Protobuf • Sqlite • Syzygy • Usrsctp • Yasm • zlib
  • 19. Chrome Specific Webrtc Bits • //src/content/renderer/media/ • //src/third_party/WebKit/public/platform/ • //src/third_party/WebKit/Source/modules/mediastream • //src/content/browser/media/ • //src/content/browser/media/capture/ • //src/media/capture/video/ • //src/content/browser/renderer_host/media/ • . • . • . • . • . • . • .
  • 20. Chrome Specific Webrtc Bits // MediaStreamManager is used to open/enumerate media capture devices (video // supported now). Call flow: // 1. GenerateStream is called when a render process wants to use a capture // device. // 2. MediaStreamManager will ask MediaStreamUIController for permission to // use devices and for which device to use. // 3. MediaStreamManager will request the corresponding media device manager(s) // to enumerate available devices. The result will be given to // MediaStreamUIController. // 4. MediaStreamUIController will, by posting the request to UI, let the // users to select which devices to use and send callback to // MediaStreamManager with the result. // 5. MediaStreamManager will call the proper media device manager to open the // device and let the MediaStreamRequester know it has been done.
  • 21. Chrome GUM example capturers / frames / Encoding / ….
  • 22. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser)GUM JS API (tab/sandbox) 1. Send request
  • 23. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser)GUM JS API (tab/sandbox) 2. Check if MST is already available
  • 24. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser)GUM JS API (tab/sandbox) Security Manager (source, origin) 3. Check rights 2. Check if MST is already available - NO
  • 25. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser) Audio Capturer Video Capturer Security Manager (source, origin) GUM JS API (tab/sandbox) 4. Ask Corresponding capturer type to start capturing 3. Check rights - OK
  • 26. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser) Audio Capturer Video Capturer Security Manager (source, origin) A GUM JS API (tab/sandbox) 4. Ask Corresponding capturer type to create one - OK V 5. Store the MST
  • 27. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser) Audio Capturer Video Capturer Security Manager (source, origin) A GUM JS API (tab/sandbox) V 6. Trigger callback Keep feeding frames
  • 28. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser) Audio Capturer Video Capturer Security Manager (source, origin) A GUM JS API (tab/sandbox) V NOTE 1: second call for same device with same constraints will directly return the MST, that allows to share streams across tabs without blocking
  • 29. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser) Audio Capturer Video Capturer Security Manager (source, origin) A GUM JS API (tab/sandbox) V NOTE 2: Recently, a second call for the same device but with different constraints (think simulcast) will indeed return a different resolution. Before it would return the first resolution asked.
  • 30. Chrome Specific Webrtc Bits Media Stream Manager (singleton@browser) Audio Capturer Video Capturer Security Manager (source, origin) A GUM JS API (tab/sandbox) V NOTE 3: Not only this allow to share cams across processes, it allows for global echo cancellation (yes, including the key strokes). Before tabs could cross feed.
  • 31. Chrome Screensharing 2 steps (1) Media Stream Manager (singleton@browser) Screen/Windows/Tab Capturer Security Manager (source, origin) Screensharing (extension) 1 2 3 4 S 5
  • 32. Chrome Screensharing 2 steps (2) Media Stream Manager (singleton@browser) Screen/Windows/Tab Capturer S GUM JS API (tab/sandbox) With ID
  • 34. Firefox Variation – Desktop 1process
  • 36. 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
  • 37. Webkit (safari ?) Current work: - Implement datachannel for webkit (yin liu) - Implement webrtc.org as a back end for webkit (adam tiamou) - Implement windows support for OWR
  • 41. Webkit Arch – Data Channel
  • 42. Webkit Arch – Data Channel

Editor's Notes

  1. https://wiki.mozilla.org/Media/WebRTC/WebRTCE10S
  2. https://wiki.mozilla.org/Media/WebRTC/WebRTCE10S