rolios / lib-testrtc

lib-testrtc: Turns TestRTC into a library to make webrtc tests embeddable in any page.

lib-testrtc

Turns TestRTC into a library to make tests embeddable in any page.

TestRTC

WebRTC troubleshooter provides a set of tests that can be easily run by a user to help diagnose WebRTC related issues. The user can then download a report containing all the gathered information or upload the log and create a temporary link with the report result.

Usage

Install with npm / yarn npm install lib-testrtc yarn add lib-testrtc

Or import the script into your page and use the global object TestRTC.

<script type="text/javascript" src="testrtc-min.js"></script>
<script type="text/javascript">
    const testRTC = new TestRTC({
      turnUsername: 'USERNAME',
      turnCredential: 'PASSWORD',
      turnURI: 'YOUR_TURN_URL',
      strunURI: 'YOUR_STUN_URI',
    });

    testRTC.start();
</script>

Callbacks

You can suscribe to different callbacks to get the tests reports:

// Test Progress
testRTC.onTestProgress((suiteName, testName, progress) => {});
// Global Progress
testRTC.onGlobalProgress((finishedCount, remainingCount) => {});
// Result
testRTC.onTestProgress((suiteName, testName, result) => {});
// Logs
testRTC.onTestReport((suite, test, level, message) => {});
// Tests completed
testRTC.onComplete(() => {});
// Tests stopped
testRTC.onStopped(() => {});

Suite control

The following commands are available:

testRTC.start() (Start from the beggining) testRTC.pause() testRTC.resume() testRTC.stop() (Stop the test suite)

testRTC.state gives an indication of the current state of the suite: stopped/started/paused.

Filtering tests

You can prevent the execution of a suites or a test case by passing it through a filter array in parameter:

const testRTC = new TestRTC(
  {
    turnUsername: 'USERNAME',
    turnCredential: 'PASSWORD',
    turnURI: 'YOUR_TURN_URL',
    strunURI: 'YOUR_STUN_URI'
  },
  (filter = [TestRTC.SUITES.CAMERA, TestRTC.TESTS.IPV6ENABLED])
);

The accepted values can be found here

Tests descriptions

• Microphone
  • Audio capture
    • Checks the microphone is able to produce 2 seconds of non-silent audio
    • Computes peak level and maximum RMS
    • Clip detection
    • Mono mic detection
• Camera
  • Check WxH resolution
    • Checks the camera is able to capture at the requested resolution for 5 seconds
    • Checks if the frames are frozen or muted/black
    • Detects how long to start encode frames
    • Reports encode time and average framerate
  • Check supported resolutions
    • Lists resolutions that appear to be supported
• Network
  • Udp/Tcp
    • Verifies it can talk with a turn server with the given protocol
  • IPv6 connectivity
    • Verifies it can gather at least one IPv6 candidate
• Connectivity
  • Relay
    • Verifies connections can be established between peers through a TURN server
  • Reflexive
    • Verifies connections can be established between peers through NAT
  • Host
    • Verifies connections can be established between peers with the same IP address
• Throughput
  • Data throughput
    • Establishes a loopback call and tests data channels throughput on the link
  • Video bandwidth
    • Establishes a loopback call and tests video performance on the link
    • Measures rtt on media channels.
    • Measures bandwidth estimation performance (rampup time, max, average)
  • Network latency
    • Establishs a loopback call and sends very small packets (via data channels) during 5 minutes plotting them to the user. It can be used to identify issues on the network.

Build lib-testrtc

yarn && grunt build

Feel free to contribute by opening a pull request.