Telestax Blog

WebRTC-test – an open source WebRTC testing tool

We are introducing the Alpha release of  WebRTC-test, an open source tool for WebRTC testing.

WebRTC-test is an open source tool used for testing client and server media components. It utilizes WebRTC, like Media Servers, WebRTC powered SIP clients, etc.

Supported  features:

  • Facilitates load testing for WebRTC media components. We successfully utilized it to load test RestcommONE Media Server in an end-to-end manner to simulate real-world scenarios.
  • Can be configured to work on headless hosts and manage WebRTC media successfully in the absence of real loudspeaker and microphone. This could make it ideal also for functional testing on CI systems.
  • Tested on Chrome and Firefox. Hosted on both OSX and GNU/Linux hosts
  • Automated RestcommONE provisioning for ease of use.
  • Implemented external service for RestcommONE so that a Webrtc client pool can be used to evenly distribute load.

Please keep in mind that the tool is a bit tied to RestcommONE for now, but the ultimate goal is to make the server side pluggable and independent, so that it can be used with any server for webrtc testing. Also, we plan to make the whole flow more streamlined and intuitive, as well as consider adding support for webrtc apps hosted in mobile devices instead of web apps only.

Project Home, Quick Start: https://github.com/RestComm/webrtc-test
Changelog: https://github.com/RestComm/webrtc-test/issues?q=milestone%3A%221.0.0+ALPHA+Release%22.

We strongly encourage everyone to jump in and play around with the code or better yet contribute to the project! You can get started by checking our open issues labeled with Help Wanted as well as share your ideas & suggestions.

Check out our RestcommONE forum and post all of your questions at: https://groups.google.com/forum/#!forum/restcomm

Get awesome content in your inbox every week.

Give it a try. It only takes a click to unsubscribe.