WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.
The official WebRTC provides native code of Android and iOS examples.
Build environment:
- Ubuntu 16.04
- Android Studio 3.x or higher
- System Python 2.x (3.x or Anaconda python may fail)
- Git
Install Prerequisite Software:
git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git
Add depot_tools to the end of your PATH (you will probably want to put this in your ~/.bashrc or ~/.zshrc ). Assuming you cloned depot_tools to /path/to/depot_tools:
$ export PATH=$PATH:/path/to/depot_tools
Get the code:
Create a working directory, enter it, and run:
fetch --nohooks webrtc_android
gclient sync
You can type gclient sync command again if code synchronizing is interrupted.
Setup the environment:
cd src/
./build/install-build-deps.sh
./build/install-build-deps-android.sh
Compile:
gn gen out/Debug --args='target_os="android" target_cpu="arm"'
You can specify a directory of your own choice instead of out/Debug, to enable managing multiple configurations in parallel.
- To build for ARM64: use target_cpu="arm64" .
- To build for 32-bit x86: use target_cpu="x86" .
- To build for 64-bit x64: use target_cpu="x64" .
ninja -C out/Debug
ninja -C out/Debug AppRTCMobile
Generate the project files:build/android/gradle/generate_gradle.py --output-directory $PWD/out/Debug --target "//examples:AppRTCMobile" --use-gradle-process-resources --split-projects --canaryIf you want to clean built files, run this:
gn clean out/Default
Import project in Android Studio:
Import the project in Android Studio. (Do not just open it.) The project is located in out/Debug/gradle. If asked which SDK to use, choose to use Android Studio's SDK. When asked whether to use the Gradle wrapper, press “OK”.Test the App:
We are going to use https://appr.tc/ to test the communication.Open it in a new tab. Next, either enter a specific string or choose random number as room id and then press JOIN. The browser will ask you for permissions of microphone and camera. Click Yes. On app side, enter the room id you are in. Press the call button to check whether or not it is working.
AppRTC parameters
You can add extra parameters to the link. Go to the AppRTC parameters for more information and usagesTutorials:
There are some tutorials with good explanation for beginners to learn WebRTC.
- WebRTC Tutorials on Youtube
- WebRTC in the real world: STUN, TURN and signaling
- Part 1: Introduction to WebRTC
- Part 2: Introduction to PeerConnection
- Part 3: Peer-to-Peer Video Calling — Loopback
- Part 4: Peer-to-Peer Video Calling with socket.io
- My Hexaville - Android WebRTC Sending Images Using DataChannel
- My Hexaville - Android WebRTC Signaling
- My Hexaville - Android WebRTC PeerConnection
- My Hexaville - Android WebRTC DataChannel Sending Text Messages
- My Hexaville - Android Video Calls Rendering Local Camera Video
- My Hexaville - Android Video Calls with WebRTC
- Google Codelab - WebRTC
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