This document discusses using C++ and the WebRTC Native APIs to build real-time communications applications. It provides an overview of the WebRTC architecture and APIs, describes how to set up the native development environment by checking out the WebRTC code from Chromium, and gives examples of using key WebRTC classes like PeerConnection, DataChannel, and observers. It also covers building for different platforms like Linux and macOS and integrating WebRTC with other libraries and protocols.

1. C++ WebRTC
(DataChannel)
/ @llamerada_jp
WebRTC Meetup Tokyo #20

2. SE
/ /
GitHub : llamerada-jp
facebook : ito.yuuji
twitter : @llamerada_jp
blog : http://llamerad-jp.hatenablog.com/

3. /
PROCESSWARP
Web /

4. /
www.oinari.app
libvein
Server

5. /
libvein
Web /
2 PubSub
KVS
Seed / Server

6. Documents /
https://webrtc.org/native-code/development/
&
API code reading
WebRTC ( )
https://gist.github.com/szktty/
a47213f0077294c64ea58621c2dcfaf2
android iOS
CUI WebRTC
https://github.com/llamerada-jp/webrtc-cpp-sample
WebRTC
https://github.com/llamerada-jp/libwebrtc

7. Native APIs

8. WebRTC Native APIs ?
WebView / Electron
WebSocket
iOS / Android (C++ )
WebRTC Native APIs
API
API

9. WebRTC API ( )
connect
SDP
ICE
send data
disconnect
event
change status
recv data
raise error

10. libuv
libuv
main Thread
rtc::Runnable Thread
webrtc::
CreatePeerConnectionFactory
rtc::Runnable::Run
uv_loop
uv_async_send Event!

11. libvein
JavaScritp
WebAssembly
WebRTC/WebSocket API
SDP / ICE (Server /
Seed )
libvein core (C++)
WebSocket
Wrapper
WebRTC
Wrapper
Native
WebRTC
Browser
WebRTC
Browser
WebSocket
WebSocket++
C++
API
C
API
Python JavaScript

12. https://webrtc.org/native-code/development/
http://commondatastorage.googleapis.com/chrome-infra-docs/flat/depot_tools/
docs/html/depot_tools_tutorial.html

13. $ cd <workdir>
$ git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git
$ export PATH=$PATH:`pwd`/depot_tools
$ cd <workdir>
$ mkdir webrtc-checkout
$ cd webrtc-checkout
$ fetch --nohooks webrtc
$ gclient sync
$ gn gen out/Default --args=‘is_debug=false’
$ ninja -C out/Default

14. Chrome WebRTC

15. @ WebRTC
Σ( ω )

16. chromium URL
https://omahaproxy.appspot.com/all
OS stable
branch_commit
HASH of chromium

17. $ cd <workdir>
$ git clone https://chromium.googlesource.com/chromium/src chromium
$ cd chromium
$ git show <HASH of chromium>:DEPS
< >
‘src/third_party/webrtc':
Var('webrtc_git') + '/src.git' + '@' + ‘784fccbd71c0130b32345d81459b5d0cb07ff6e5',
< >
HASH of webrtc

18. < gclient sync >
$ git fetch
$ git checkout <HASH of webrtc>
$ gclient sync
$ gn gen out/Default [--args='is_debug=false']
$ ninja -C out/Default [<target>]

19. @ Linux
static library(.a ) .o
.a
ar 1 .a
src/out/Default/obj/examples/peerconnection_client.ninja
build <target>: link <obj1> <obj2> …
build ./peerconnection_client: link obj/examples/
peerconnection_client/conductor.o obj/examples/
peerconnection_client/defaults.o obj/examples/
peerconnection_client/peer_connection_client.o obj/
examples/peerconnection_client/main.o obj/examples/
peerconnection_client/main_wnd.o obj/api/video/
video_frame_i420/i420_buffer.o obj/rtc_base/checks/
checks.o obj/rtc_base/stringutils/string_to_number.o

20. @ macOS
obj/sdk/mac_framework_objc_shared_library/WebRTC
obj/api/video_codecs/libbuiltin_video_encoder_factory.a
obj/api/video_codecs/libbuiltin_video_decoder_factory.a
src/out/Default/obj/sdk/mac_framework_objc_shared_library.ninja
src/obj/api/video_codecs/libbuiltin_video_encoder_factory.a
src/obj/api/video_codecs/libbuiltin_video_decoder_factory.a

21. nm
https://github.com/llamerada-jp/search-
symbol
chromium

22. peerconnection_client
https://github.com/llamerada-jp/webrtc-cpp-sample
API JavaScript JavaScript
C++ API

23. PeerConnectionObserver
class PCO : public webrtc::PeerConnectionObserver {
void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state) override;
void OnAddStream(rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
void OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
// DataChannel !
void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
void OnRenegotiationNeeded() override;
void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
// ICE
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
};

24. DataChannelObserver
class DCO : public webrtc::DataChannelObserver {
//
void OnStateChange() override;
//
void OnMessage(const webrtc::DataBuffer& buffer) override;
void OnBufferedAmountChange(uint64_t previous_amount) override;
};

25. SessionDescriptionObserver
class CSDO : public webrtc::CreateSessionDescriptionObserver {
void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
void OnFailure(const std::string& error) override;
};
class SSDO : public webrtc::SetSessionDescriptionObserver {
void OnSuccess() override;
void OnFailure(const std::string& error) override;
};
rtc::scoped_refptr<CSDO> csdo (new rtc::RefCountedObject<CSDO>());
rtc::scoped_refptr<SSDO> ssdo (new rtc::RefCountedObject<SSDO>());

26. &
CreatePeerConnectionFactory
class CustomRunnable : public rtc::Runnable {
public:
void Run(rtc::Thread* subthread) override {
peer_connection_factory = webrtc::CreatePeerConnectionFactory(
nullptr /* network_thread */, nullptr /* worker_thread */,
nullptr /* signaling_thread */, nullptr /* default_adm */,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(),
nullptr /* audio_mixer */, nullptr /* audio_processing */);
if (peer_connection_factory.get() == nullptr) {
// Error
}
subthread->Run();
}
};

27. rtc::PhysicalSocketServer socket_server;
thread.reset(new rtc::Thread(&socket_server));
rtc::InitializeSSL();
CustomRunnable runnable;
thread->Start(&runnable);
< peer_connection_factory >
peer_connection = peer_connection_factory->CreatePeerConnection(configuration, nullptr, nullptr, &pco);
webrtc::DataChannelInit config;
// DataChannel
data_channel = peer_connection->CreateDataChannel("data_channel", &config);
data_channel->RegisterObserver(&dco);
peer_connection->CreateOffer(csdo, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());

28. std::string message(“hello!!”,);
webrtc::DataBuffer buffer(
rtc::CopyOnWriteBuffer(
message.c_str(),
message.size()),
true);
data_channel->Send(buffer);

29. class DCO : public webrtc::DataChannelObserver {
//
void OnMessage(const webrtc::DataBuffer& buffer) override {
std::cout << std::string(buffer.data.data<char>(), buffer.data.size()) << std::endl;
}
};

30. // Close
peer_connection->Close();
peer_connection = nullptr;
data_channel = nullptr;
peer_connection_factory = nullptr;
thread->Quit();
thread.reset();
rtc::CleanupSSL();

31. macOS : WEBRTC_MAC=1 WEBRTC_POSIX=1
Linux : WEBRTC_LINUX=1 WEBRTC_POSIX=1
clang (libc++ )
-stdlib=libc++
C++ RTTI -fno-rtti
typeinfo dynamic_cast
--start-group --end-group

32. $ clang++ -I<workdir/webrtc-checkout/src> -I<workdir/webrtc-checkout/src/
third_party/abseil-cpp> -DWEBRTC_MAC=1 -DWEBRTC_POSIX=1 -std=c++11 -W -Wall -
Wno-unused-parameter -O0 -g -fno-rtti -o <output> -o <source>
(github ) build
./build.sh
$ cd <path to build>
$ make clean
$ make VERBOSE=1

33. OpenSSL BoringSSL API
HTTP WebSocket
BoringSSL
WebRTC Third party src/third_party

34. WebRTC BSD
src/third_party
OSS WebRTC

35. WebRTC C++