https://github.com/qxip/webrtc-test-alpine
WebRTC Chromium RTC Quality Tester w/ PaStash output
https://github.com/qxip/webrtc-test-alpine
Last synced: about 1 year ago
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WebRTC Chromium RTC Quality Tester w/ PaStash output
- Host: GitHub
- URL: https://github.com/qxip/webrtc-test-alpine
- Owner: QXIP
- License: agpl-3.0
- Created: 2021-05-21T15:10:25.000Z (about 5 years ago)
- Default Branch: main
- Last Pushed: 2023-03-06T02:39:06.000Z (over 3 years ago)
- Last Synced: 2025-04-07T20:21:23.665Z (about 1 year ago)
- Language: JavaScript
- Homepage:
- Size: 4.04 MB
- Stars: 8
- Watchers: 2
- Forks: 1
- Open Issues: 15
-
Metadata Files:
- Readme: README.md
- License: LICENSE
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README
# WebRTC stress test
A tool for running concurrent WebRTC sessions using chromium web browser in headless mode.
Components used:
- NodeJS application.
- Puppeteer library for controlling chromium instances.
- RTC stats logging with [paStash](https://github.com/sipcapture/paStash).
## Configuration options
See the [config documentation](CONFIG.md).
The `DEBUG_LEVEL` environment is used for enabled debug messages; see [debug-level](https://github.com/commenthol/debug-level#readme) for syntax.
## Statistics
Example output:
```
-- Mon, 08 Mar 2021 11:41:48 GMT -------------------------------------------------------------------
name count sum mean stddev 25p min max
cpu 1 67.66 67.66 0.00 67.66 67.66 67.66 %
memory 1 801.13 801.13 0.00 801.13 801.13 801.13 MB
tabs 1 1 1 0 1 1 1
-- Inbound audio -----------------------------------------------------------------------------------
received 1 0.02 0.02 0.00 0.02 0.02 0.02 MB
rate 1 0.53 0.53 0.00 0.53 0.53 0.53 Kbps
lost 1 0.00 0.00 0.00 0.00 0.00 %
jitter 1 0.00 0.00 0.00 0.00 0.00 s
avgJitterBufferDelay 1 85.27 0.00 85.27 85.27 85.27 ms
-- Inbound video -----------------------------------------------------------------------------------
received 2 26.62 13.31 13.22 13.31 0.09 26.53 MB
rate 2 838.72 419.36 411.30 419.36 8.06 830.66 Kbps
lost 1 0.00 0.00 0.00 0.00 0.00 %
jitter 1 1.88 0.37 1.77 0.98 2.65 s
avgJitterBufferDelay 1 90.86 0.00 90.86 90.86 90.86 ms
width 1 1280 0 1280 1280 1280 px
height 1 720 0 720 720 720 px
-- Outbound audio ----------------------------------------------------------------------------------
sent 1 0.50 0.50 0.00 0.50 0.50 0.50 MB
retransmitted 1 0.00 0.00 0.00 0.00 0.00 0.00 MB
rate 1 0.00 0.00 0.00 0.00 0.00 0.00 Kbps
-- Outbound video ----------------------------------------------------------------------------------
sent 3 43.62 14.54 7.70 10.06 4.68 23.49 MB
retransmitted 3 0.00 0.00 0.00 0.00 0.00 0.00 MB
rate 3 0.00 0.00 0.00 0.00 0.00 0.00 Kbps
qualityLimitResolutionChanges 3 0 0 0 0 0 0
width 1 1280 0 1280 1280 1280 px
height 1 720 0 720 720 720 px
fps 1 25 0 25 25 25 fps
```
Statistics values:
| Name | Count | Desscription |
| :------------------------ | :----------- | :----------- |
| cpu | Total sessions | The browser process cpu usage. |
| memory | Total sessions | The browser process memory usage. |
| tabs | Total sessions | The browser current opened tabs. |
| received | Total inbound streams | The `bytesReceived` value for each stream. |
| sent | Total outbound streams | The `bytesSent` value for each stream. |
| retransmitted | Total outbound streams | The `retransmittedBytesSent` value for each stream. |
| rate | Total streams | The stream bitrate. |
| lost | Total streams | The stream [lost packets](https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetslost) %. |
| jitter | Total streams | The stream [jitter](https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter) in seconds. |
| avgJitterBufferDelay | Total decoded tracks | The inbound average [jitter buffer delay](https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay). |
| qualityLimitResolutionChanges | Total outbound video streams | The `qualityLimitationResolutionChanges` [value](https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges) for each outbound video stream. |
| width | Total sent or received videos | The sent or received video width. |
| height | Total sent or received videos | The sent or received video height. |
| fps | Total sent | The sent video frames per second. |
## Examples
See the 'examples' folder
### Mediasoup demo
Starts one send-receive participant:
```sh
docker pull qxip/webrtc-stress-test-pastash:latest
docker run -it --rm --name=webrtc-stress-test-publisher \
-v /dev/shm:/dev/shm \
-e VIDEO_PATH=/app/video.mp4 \
-e URL=$MEDIASOUP_DEMO_URL \
-e URL_QUERY='roomId=test&displayName=Publisher-$s-$t' \
-e SESSIONS=1 \
-e TABS_PER_SESSION=1 \
-e SHOW_STATS=false \
-e ENABLE_RTC_STATS=true \
-e PASTASH_WSS='wss://pastash.collector:9090' \
-e PASTASH_ID='my_test_123' \
qxip/webrtc-stress-test-pastash:latest
```
Starts 10 receive-only participants:
```sh
docker pull qxip/webrtc-stress-test-pastash:latest
docker run -it --rm --name=webrtc-stress-test-viewer \
-v /dev/shm:/dev/shm \
-e URL=$MEDIASOUP_DEMO_URL \
-e URL_QUERY='roomId=test&displayName=Viewer-$s-$t&produce=false' \
-e SESSIONS=1 \
-e TABS_PER_SESSION=10 \
-e SHOW_STATS=false \
-e ENABLE_RTC_STATS=true \
-e PASTASH_WSS='wss://pastash.collector:9090' \
-e PASTASH_ID='my_test_123' \
qxip/webrtc-stress-test-pastash:latest
```
### Meething
Starts one send-receive participant, with a random id:
```sh
docker pull qxip/webrtc-stress-test-pastash:latest
docker run -it --rm --name=webrtc-stress-test-publisher \
-v /dev/shm:/dev/shm \
-e VIDEO_PATH=/app/video.mp4 \
-e URL=$MEETHING_URL \
-e SCRIPT_PATH=/app/scripts/meething.js \
-e SESSIONS=1 \
-e TABS_PER_SESSION=1 \
-e SHOW_STATS=false \
-e ENABLE_RTC_STATS=true \
-e PASTASH_WSS='wss://pastash.collector:9090' \
-e PASTASH_ID='my_test_123' \
qxip/webrtc-stress-test-pastash:latest
```
### Jitsi
Starts one send-receive participant:
```sh
docker pull qxip/webrtc-stress-test-pastash:latest
docker run -it --rm --name=webrtc-stress-test-publisher \
-v /dev/shm:/dev/shm \
-e VIDEO_PATH=/app/video.mp4 \
-e URL=$JITSI_ROOM_URL \
-e URL_QUERY='#config.prejoinPageEnabled=false&userInfo.displayName=Participant-$s-$t' \
-e SESSIONS=1 \
-e TABS_PER_SESSION=1 \
-e SHOW_STATS=false \
-e ENABLE_RTC_STATS=true \
-e PASTASH_WSS='wss://pastash.collector:9090' \
-e PASTASH_ID='my_test_123' \
qxip/webrtc-stress-test-pastash:latest
```
## Running from source code
```sh
git clone https://github.com/qxip/webrtc-stress-test-pastash.git
cd webrtc-stress-test
# build the chromium customized version
# cd chromium
# ./build.sh setup
# ./build.sh apply_patch
# ./build.sh build
# install the package (on Ubuntu/Debian)
# dpkg -i ./chromium-browser-unstable_-1_amd64.deb
# cd ..
# sendrecv test
DEBUG_LEVEL=DEBUG:* yarn start:dev \
--url=https://127.0.0.1:3443/test \
--url-query='displayName=SendRecv $s/$S-$t/$T' \
--video-path=./video.mp4 \
--script-path=./scripts/edumeet-sendrecv.js \
--sessions=1 \
--tabs-per-session=1 \
--enable-page-log=true
# recv only
DEBUG_LEVEL=DEBUG:* yarn start:dev \
--url=https://127.0.0.1:3443/test \
--url-query='displayName=Recv $s/$S-$t/$T' \
--script-path=./scripts/edumeet-recv.js \
--sessions=1 \
--tabs-per-session=10 \
--enable-page-log=true \
--use-null-video-decoder=true
```
## Setup
Install pastash & the required plugins using npm
```
npm install -g @pastash/pastash @pastash/output_loki
```
---------------
## Pastash Collector Recipe
The following recipe will provide a websocket receiver for webRTC statistics on port `9090` and shipping to Grafana Loki
Create and store this on your filesystem, ie: `/some/path/to/pastash.conf` and add in your `grafana` account details:
```
input {
ws {
host => 0.0.0.0
port => 9090
ssl => true
ssl_key => "/etc/letsencrypt/live/your.domain/privkey.pem"
ssl_cert => "/etc/letsencrypt/live/your.domain/cert.pem"
}
}
output {
loki {
basic_auth_password => "some_very_secure_password_hash_here"
basic_auth_user => "1234"
host => "logs-us-west1.grafana.net"
port => 80
path => "/api/prom/push"
}
}
```
Once you're ready, run your paStash socket manually or using pm2
```
pastash --config_file=/some/path/to/pastash.conf
```
### Credits
* Based on [webrtc-stress-test](https://github.com/vpalmisano)