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.

Automatic tests

• 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)

Manual tests

Due to their time duration they are not part of the normal test suite and need to be run explicitly.

• [Network latency](https://test.webrtc.org/?test_filter=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.

Other manual test pages

• Audio and Video streams
• Iframe apprtc
• Iframe video
• Multiple audio streams
• Multiple peerconnections
• Multiple video devices
• Multiple video streams
• Peer2peer
• Peer2peer iframe
• Single audio stream
• Single video stream

Contributing

Patches and issues welcome! See CONTRIBUTING for instructions. All contributors must sign a contributor license agreement before code can be accepted. Please complete the agreement for an individual or a corporation as appropriate. The Developer's Guide for this repo has more information about code style, structure and validation.

Development

Make sure to install NodeJS and NPM before continuing. Note that we have been mainly been using Posix when developing TestRTC hence developer tools might not work correctly on Windows.

Install developer tools and frameworks

npm install

Install dependencies

bower update

Run linters (currently very limited set is run)

grunt

Build testrtc

Cleans out/ folder if it exists else it's created, then it copies and vulcanizes the resources needed to deploy this on Google App Engine.

grunt build

Run non vulcanized version of TestRTC using Google App Engine SDK for Python. This is useful while developing.

python dev_appserver.py app.yml

Run vulcanized version of TestRTC using Google App Engine SDK for Python (Requires the Build testrtc step to be performed first).

python dev_appserver.py out/app.yml