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The currently supported platforms are Windows, Mac OS X, Linux, Android and iOS. See the Android and iOS pages for build instructions and example applications specific to these mobile platforms.

Before You Start

First, be sure to install the prerequisite software.

Getting the Code

For desktop development:

  1. Create a working directory, enter it, and run fetch webrtc:

    mkdir webrtc-checkout
    cd webrtc-checkout
    fetch --nohooks webrtc
    gclient sync

    NOTICE: During your first sync, you’ll have to accept the license agreement of the Google Play Services SDK.

    The checkout size is large due the use of the Chromium build toolchain and many dependencies. Estimated size:

    • Linux: 6.4 GB.
    • Linux (with Android): 16 GB (of which ~8 GB is Android SDK+NDK images).
    • Mac (with iOS support): 5.6GB
  2. Optionally you can specify how new branches should be tracked:

    git config branch.autosetupmerge always
    git config branch.autosetuprebase always
  3. Alternatively, you can create new local branches like this (recommended):

    cd src
    git checkout master
    git new-branch your-branch-name

See Android and iOS pages for separate instructions.

Updating the Code

Update your current branch with:

git pull

NOTICE: if you’re not on a branch, git pull won’t work, and you’ll need to use git fetch instead.

Periodically, the build toolchain and dependencies of WebRTC are updated. To get such updates you must run:

gclient sync


Ninja is the default build system for all platforms.

See Android and iOS for build instructions specific to those platforms.

Generating Ninja project files

Ninja project files are generated using GN. They’re put in a directory of your choice, like out/Debug or out/Release, but you can use any directory for keeping multiple configurations handy.

To generate project files using the defaults (Debug build), run (standing in the src/ directory of your checkout):

gn gen out/Default

NOTICE: Debug builds are component builds (shared libraries) by default unless is_component_build=false is passed to gn gen --args. Release builds are static by default.

To generate ninja project files for a Release build instead:

gn gen out/Default --args='is_debug=false'

To clean all build artifacts in a directory but leave the current GN configuration untouched (stored in the file), do:

gn clean out/Default

See the GN documentation for all available options. There are also more platform specific tips on the Android and iOS pages.


When you have Ninja project files generated (see previous section), compile (standing in src/) using:

For Ninja project files generated in out/Default:

ninja -C out/Default

Using Another Build System

Other build systems are not supported (and may fail), such as Visual Studio on Windows or Xcode on OSX. GN supports a hybrid approach of using Ninja for building, but Visual Studio/Xcode for editing and driving compilation.

To generate IDE project files, pass the --ide flag to the GN command. See the GN reference for more details on the supported IDEs.

Working with Release Branches

To see available release branches, run:

git branch -r

NOTICE: If you only see your local branches, you have a checkout created before our switch to Git (March 24, 2015). In that case, first run:

cd /path/to/webrtc/src
gclient sync --with_branch_heads
git fetch origin

You should now have an entry like this under [remote “origin”] in .git/config:

fetch = +refs/branch-heads/*:refs/remotes/branch-heads/*

To create a local branch tracking a remote release branch (in this example, the 43 branch):

git checkout -b my_branch refs/remotes/branch-heads/43

Commit log for the branch:

To browse it:

For more details, read Chromium’s Working with Branches and Working with Release Branches pages.

Contributing Patches

Please see Contributing Fixes for information on how to get your changes included in the WebRTC codebase. You’ll also need to setup authentication for committing, below.

Committing Code

To commit code directly to the Git repo, you have to be a committer. CLs created by external contributors can be committed via the Commit Queue (CQ).

The source of truth is the Git repository at To be able to push commits to it, you need to perform the steps below (assuming you’re a committer).

If you already have a .netrc/.gitcookies file (most Chromium committers already do), you can skip steps 1 and 2.

  1. Go to and login with your account.

  2. Follow the instructions on how to store the credentials in the .gitcookies file in your home directory.

  3. Go to and login with your account. This will create the user in the Gerrit permission system so it can be added to the right committers group.

  4. Ask to be added to the committers group to get push access.

  5. Make sure you have set the and Git config settings as specified at the depot tools setup page. If you’re also a Chromium committer, read the next section.

Commit a change list to the Git repo using:

git cl land

NOTICE: On Windows, you’ll need to run this in a Git bash shell in order for gclient to find the .gitcookies file.

Sometimes it’s necessary to bypass the presubmit checks (like when fixing an error that has closed the tree). Then use the --bypass-hooks flag.

Chromium Committers

Many WebRTC committers are also Chromium committers. To make sure to use the right account for pushing commits to WebRTC, use the Git config setting. The recommended way is to have the account set globally as described at the depot tools setup page and then set locally for the WebRTC repos using (change to your address):

cd /path/to/webrtc/src
git config

Example Applications

WebRTC contains several example applications, which can be found under src/webrtc/examples and src/talk/examples. Higher level applications are listed first.


Peerconnection consist of two applications using the WebRTC Native APIs:

  • A server application, with target name peerconnection_server

  • A client application, with target name peerconnection_client (not currently supported on Mac/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 (by 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.

Testing peerconnection_server

Start an instance of peerconnection_server application.

Open src/webrtc/examples/peerconnection/server/server_test.html in your browser. Click Connect. Observe that the peerconnection_server announces your connection. Open one more tab using the same page. Connect it too (with a different name). It is now possible to exchange messages between the connected peers.

Call App

Target name call (currently disabled). An application that establishes a call using libjingle. 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 interoperate with Google Talk Video. If you use other input RTP dump, you may need to change the codecs in (lines 215-222).

Relay Server

Target name relayserver. Relays traffic when a direct peer-to-peer connection can’t be established. Can be used with the call application above.

STUN Server

Target name stunserver. Implements the STUN protocol for Session Traversal Utilities for NAT as documented in RFC 5389.

TURN Server

Target name turnserver. In active development to reach compatibility with RFC 5766.