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WebRTC audio/video call and conferencing server.

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Open-RTC

Open-RTC implements a WebRTC audio/video call and conferencing server and web client.

The latest source of Open-RTC can be found on GitHub.

(original project was https://github.com/strukturag/spreed-webrtc)

Build prerequisites

Runtime dependencies

Open-RTC compiles directly to native code and has no external runtime dependencies. See here for details.

Building

Build Status

If you got open-rtc from the git repository, you will first need to run the included autogen.sh script to generate the configure script.

Configure does try to find all the tools on your system at the standard locations. If the dependencies are somewhere else, add the corresponding parameters to the ./configure call.

$ ./configure
$ make

On FreeBSD, the default make has a different syntax, so gmake must be used there.

Build separately

There are several make targets available. Get Go external dependencies at least once with make get, all the other targets depend on this.

$ make get
$ make assets
$ make binary

Server startup

open-rtc-server [OPTIONS]

Options

Usage of open-rtc-server:
  -c string
        Configuration file. (default "./server.conf")
  -cpuprofile string
        Write cpu profile to file.
  -dc string
        Default configuration file.
  -h    Show this usage information and exit.
  -l string
        Log file, defaults to stderr.
  -memprofile string
        Write memory profile to this file.
  -oc string
        Override configuration file.
  -v    Display version number and exit.

An example configuration file can be found in server.conf.in.

Usage

Connect to the server URL and port with a web browser and the web client will launch.

Development

To build styles and translations, further dependencies are required. The source tree contains already built styles and translations, so these are only required if you want to make changes.

The following Node.js modules are required, these may be installed locally by running npm install from the project root. Consult the package.json file for more details.

Styles can be found in src/styles. Translations are found in src/i18n. Each folder has its own Makefile to build the corresponding files. Check the Makefile.am templates for available make targets.

Javascript console logging is automatically disabled and can be enabled by adding the query parameter debug to your url https://my_url?debug.

Running server for development

Copy the server.conf.in to server.conf.

Build styles, javascript and binary using make. Then run ./open-rtc-server

The server runs on http://localhost:8080/ per default.

HTML changes and Go rebuilds need a server restart. Javascript and CSS reload directly.

Running for production

Open-RTC should be run through a SSL frontend proxy with support for Websockets. Example configuration for Nginx can be found in doc/NGINX.txt.

In addition, for real world use, one also needs a STUN/TURN server configured (with shared secret support).

See https://github.com/coturn/coturn for a free open source TURN server implementation. Make sure to use a recent version (we recommend 3.2). Versions below 2.5 are not supported.

For WebRTC usage, be sure to enable long-term credentials, fingerprinting, and set the realm. See https://github.com/coturn/coturn/wiki/turnserver#webrtc-usage for more information.

Running with Docker

We provide official Docker images at https://hub.docker.com/r/spreed/webrtc/. Of course you can build the Docker image yourself as well. Check the Dockerfiles in this repository for details and instructions.

Use the following command to run a Open-RTC Docker container with the default settings from our official Open-RTC Docker image.

docker run --rm --name my-open-rtc -p 8080:8080 -p 8443:8443 -v `pwd`:/srv/extra -i -t spreed/webrtc

Contributing

  1. "Fork" develop branch.
  2. Create a feature branch.
  3. Make changes.
  4. Do your commits (run make fmt and make jshint before commit).
  5. Send "pull request" for develop branch.

License

Open-RTC uses the AGPL license, see our LICENSE file.

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WebRTC audio/video call and conferencing server.

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License

AGPL-3.0, Unknown licenses found

Licenses found

AGPL-3.0
LICENSE
Unknown
COPYING

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