Ecosyste.ms: Awesome

An open API service indexing awesome lists of open source software.

https://github.com/webrtc/testrtc

WebRTC Troubeshooter PROJECT IS ON HOLD
https://github.com/webrtc/testrtc

Last synced: 2 months ago
JSON representation

WebRTC Troubeshooter PROJECT IS ON HOLD

Lists

README 

        
[![Build Status](https://travis-ci.org/webrtc/testrtc.svg)](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

```