First, be sure to install the prerequisite software.
The currently supported platforms are Windows, Mac OS X, Linux and Android.
Create a working directory, enter it, and run:
$ gclient config http://webrtc.googlecode.com/svn/trunk
Android specific steps. *
Select build system (optional for all OSs except Android where ninja is mandatory). *
Starred (*) items are described in their own section below and should be performed, if at all, in place.
If you're a committer, substitute https for http.
On Windows, use gclient.bat instead (or prefix the commands by invoking python).
Android requires that you build on a Linux machine.
The gclient sync command fetches dependencies and generates native build files for your environment using gyp (Windows: ninja/Visual Studio, OS X: ninja/XCode, Linux: ninja/make, Android: ninja). Ninja is the default build system for all platforms. It is possible to just generate new build files by calling:
Android Specific Steps
If building for Android these steps should be inlined above.
$ echo "target_os = ['android', 'unix']" >> .gclient
$ gclient sync --nohooks
$ cd trunk
$ source ./build/android/envsetup.sh
$ JAVA_HOME=<location of Java SE 6 - JDK>
Select build system
You can directly specify which build system to use. This can be done if you don't want to use ninja. Set the GYP_GENERATORS environment variable to the string:
make for Makefiles
msvs for Visual Studio
msvs-ninja for Visual Studio project building with ninja
xcode for Xcode
Note, when the build environment is set to generate Visual Studio project files, gyp will by default, generate a project for the latest version of Visual Studio installed on your computer. It is possible to specify the desired Visual Studio version as described below:
Set environment variable
$ webrtc/build/gyp_webrtc -G msvs_version=<version>
<version> is on the form YYYY.
Binaries are by default (i.e. when building with ninja) generated in
Please see Contributing bug fixes for information on how to get your changes included in the webrtc codebase.
WebRTC contains several example applications which can be found under trunk/webrtc/examples and trunk/talk/examples. Higher level applications are listed first.
Peerconnection (Application using WebRTC Native APIs)
Peerconnection consist of two applications. A server application, with target name peerconnection_server, and a client application, with target name peerconnection_client. Note that we currently don't support peerconnection_client for Mac and Android.
The client application has simple voice and video capabilities. The server enables client applications to initiate a call between clients by managing signaling messages generated by the clients.
Setting up P2P calls between peerconnection_clients
Start peerconnection_server. You should see the following message indicating that it is running:
Server listening on port 8888
Start any number of peerconnection_clients and connect them to the server. The client UI consists of a few parts:
Connecting to a server: when the application is started you must specify which machine (IP-address) the server application is running on. Once that is done you can press "Connect" or the return button.
Select a peer: once successfully connected to a server you can connect to a peer by double clicking or select+press return on a peer's name.
Video chat: when a peer has been successfully connected to, a Video chat will be displayed in full window.
Ending chat session: press Esc. You will now be back to selecting a peer.
Ending connection: press Esc and you will now be able to select which server to connect to.
Start an instance of peerconnection_server application.
Target name call (currently disabled). Call uses xmpp (as opposed to SDP used by WebRTC) to allow you to login using your gmail account and make audio/video calls with your gmail friends. It is built on top of libjingle to provide this functionality.
Further, you can specify input and output RTP dump for voice and video. It provides two samples of input RTP dump: voice.rtpdump which contains a stream of single channel, 16Khz voice encoded with G722, and video.rtpdump which contains a 320x240 video encoded with H264 AVC at 30 frames per second. The provided samples will inter-operate with Google Talk Video. If you use other input RTP dump, you may need to change the codecs in call_main.cc (lines 215 - 222).
Target name WebRTCDemo. This app does not use WebRTC native APIs. It can send and receive media streams if manually connected to another WebRTCDemo that is directly accessible (e.g. firewalls might prevent you from establishing a connection). Further it allows setting, enabling and disabling audio and video processing functionality (e.g. echo cancellation, NACK, audio codec and video codec).
Relay server (specialized server application that can be used with Call)
Target name relayserver. Relays traffic when a direct peer-to-peer connection can't be established.
Target name stunserver. Implements the STUN protocol for Session Traversal Utilities for NAT as documented in rfc5389.
Target name turnserver. In active development to reach compatibility with rfc5766.