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https://github.com/webrtc/testrtc
WebRTC Troubeshooter PROJECT IS ON HOLD
https://github.com/webrtc/testrtc
Last synced: 3 months ago
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WebRTC Troubeshooter PROJECT IS ON HOLD
- Host: GitHub
- URL: https://github.com/webrtc/testrtc
- Owner: webrtc
- License: bsd-3-clause
- Created: 2015-01-29T15:30:24.000Z (almost 10 years ago)
- Default Branch: master
- Last Pushed: 2024-03-15T13:30:11.000Z (8 months ago)
- Last Synced: 2024-04-14T09:31:27.556Z (7 months ago)
- Language: JavaScript
- Homepage: https://test.webrtc.org/
- Size: 5.23 MB
- Stars: 470
- Watchers: 51
- Forks: 212
- Open Issues: 87
-
Metadata Files:
- Readme: README.md
- Contributing: CONTRIBUTING.md
- License: LICENSE.md
Awesome Lists containing this project
- awesome-rtc - WebRTC Troubleshooter - Self-hosted one stop client-side WebRTC troubleshooter. (Operations / Monitoring)
README
[](https://travis-ci.org/webrtc/testrtc)
# TestRTC #
[WebRTC troubleshooter](https://test.webrtc.org/) 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%20latency)
* 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.
## Contributing ##
Pull requests and issues welcome! See [CONTRIBUTING](https://github.com/GoogleChrome/webrtc/blob/master/CONTRIBUTING.md) for instructions. All contributors must sign a contributor license agreement before code can be accepted. Please complete the agreement for an [individual](https://developers.google.com/open-source/cla/individual) or a [corporation](https://developers.google.com/open-source/cla/corporate) as appropriate. The [Developer's Guide](https://bit.ly/webrtcdevguide) 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 ####
```bash
npm install
```
#### Install dependencies ####
```bash
bower update
```
#### Run linters (currently very limited set is run) ####
```bash
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 vulcanized version of TestRTC using [Google App Engine SDK for Python](https://cloud.google.com/appengine/downloads) (requires the Build testrtc step to be performed first). ####
```bash
python dev_appserver.py out/app.yml
```