- Before You Start
- Getting the Code
- Updating the Code
- Working with Release Branches
- Contributing Patches
- Example Applications
Before You Start
First, be sure to install the prerequisite software.
Getting the Code
For desktop development:
Create a working directory, enter it, and run
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
Optionally you can specify how new branches should be tracked:
git config branch.autosetupmerge always git config branch.autosetuprebase always
Alternatively, you can create new local branches like this (recommended):
cd src git checkout master git new-branch your-branch-name
NOTICE: if you get
Remote: Daily bandwidth rate limit exceeded for <ip>,
make sure you’re logged in. The quota is much larger for logged in users.
Updating the Code
Update your current branch with:
git checkout master git pull origin master gclient sync git checkout my-branch git merge master
Ninja is the default build system for all platforms.
Generating Ninja project files
To generate project files using the defaults (Debug build), run (standing in the src/ directory of your checkout):
gn gen out/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 args.gn file), do:
gn clean out/Default
When you have Ninja project files generated (see previous section), compile
For Ninja project files generated in
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.
Working with Release Branches
To see available release branches, run:
git branch -r
To create a local branch tracking a remote release branch (in this example, the branch corresponding to Chrome M80):
git checkout -b my_branch refs/remotes/branch-heads/3987 gclient sync
NOTICE: depot_tools are not tracked with your checkout, so it’s possible gclient sync will break on sufficiently old branches. In that case, you can try using an older depot_tools:
which gclient # cd to depot_tools dir # edit update_depot_tools; add an exit command at the top of the file git log # find a hash close to the date when the branch happened git checkout <hash> cd ~/dev/webrtc/src gclient sync # When done, go back to depot_tools, git reset --hard, run gclient again and # verify the current branch becomes REMOTE:origin/master
The above is untested and unsupported, but it might help.
Commit log for the branch: https://webrtc.googlesource.com/src/+log/branch-heads/3987
To browse it: https://webrtc.googlesource.com/src/+/branch-heads/3987
For more details, read Chromium’s Working with Branches and Working with Release Branches pages. To find the branch corresponding to a Chrome release check the [Chromium Dashboard][https://chromiumdash.appspot.com/branches].
Please see Contributing Fixes for information on how to run
git cl upload, getting your patch reviewed, and getting it submitted.
This also includes information on how to run tryjobs, if you’re a committer.
Many WebRTC committers are also Chromium committers. To make sure to use the
right account for pushing commits to WebRTC, use the
user.email Git config
setting. The recommended way is to have the chromium.org account set globally
as described at the depot tools setup page and then set
locally for the WebRTC repos using (change to your webrtc.org address):
cd /path/to/webrtc/src git config user.email email@example.com
WebRTC contains several example applications, which can be found under
src/talk/examples. Higher level applications are
Peerconnection consist of two applications using the WebRTC Native APIs:
A server application, with target name
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
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.
Start an instance of
src/webrtc/examples/peerconnection/server/server_test.html in your
browser. Click Connect. Observe that the
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
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
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
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
relayserver. Relays traffic when a direct peer-to-peer
connection can’t be established. Can be used with the call application above.
stunserver. Implements the STUN protocol for Session Traversal
Utilities for NAT as documented in RFC 5389.
turnserver. Used for unit tests.