Webrtc Stress Test Save

WebRTC performance and quality evaluation tool.

Project README

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WebRTC Perf

GitHub page | Documentation

WebRTC performance and quality evaluation tool. It allows to validate the audio/video quality and the client CPU/memory usage when multiple connections join the same WebRTC service.

Main features:

  • A NodeJS application/library using Puppeteer for controlling chromium instances.
  • It can be executed:
    • using the pre built Docker image; this is the suggested way to run the tool without installing any dependency;
    • from sources (using git pull or npm install);
    • using the pre built executables generated for each platform.
  • It allows to inject custom Javascript source files that will run into the browser page context for automating some tasks (e.g. pressing a button to join a conference room).
  • It allows to throttle the networking configuration, limiting the ingress/egress available bandwidth, the RTT or the packet loss %.
  • It uses a patched version of chromium (see ./chromium directory) that allows to disable the video decoding, lowering the CPU requirements when running multiple browser sessions.
  • It contains an RTC stats logging module that allows to collect metrics and send them to a Prometheus Pushgateway server for live visualization with Grafana.
  • It allows to override getUserMedia and getDisplayMedia calls.
  • It allows to define alert rules and generate reports.

Install

The tool can be executed from sources, using the pre built executables or using the Docker image.

Using Npm:

echo '@vpalmisano:registry=https://npm.pkg.github.com' >> ~/.npmrc

npm install -g @vpalmisano/webrtcperf

# Install FFMpeg:
sudo apt install ffmpeg # Linux
# or:
brew install ffmpeg # MacOS

# Run a Jitsi test:
webrtcperf \
    --url="https://meet.jit.si/${JITSI_ROOM_URL}#config.prejoinPageEnabled=false" \
    --display='' \
    --show-page-log=false
# Press <q> to stop.

Using Docker:

docker pull ghcr.io/vpalmisano/webrtcperf
docker run -it --rm \
    -v /dev/shm:/dev/shm \
    ghcr.io/vpalmisano/webrtcperf \
    --url="https://meet.jit.si/$JITSI_ROOM_URL#config.prejoinPageEnabled=false" \
    --show-page-log=false \
    --sessions=1 \
    --tabs-per-session=1

Stop the tool pressing q (normal browser close) or x (it will close the process immediately).

Configuration options

See the config documentation.

Statistics

Example output:

-- Mon, 06 Feb 2023 20:46:34 GMT -------------------------------------------------------------------
                          name    count      sum     mean   stddev       5p      95p      min      max
                    System CPU        1             15.89     0.00    15.89    15.89    15.89    15.89 %
                    System GPU        1              0.00     0.00     0.00     0.00     0.00     0.00 %
                 System Memory        1             72.18     0.00    72.18    72.18    72.18    72.18 %
                      CPU/page        1    84.42    84.42     0.00    84.42    84.42    84.42    84.42 %
                   Memory/page        1  1206.90  1206.90     0.00  1206.90  1206.90  1206.90  1206.90 MB
                         Pages        1        1        1        0        1        1        1        1
                        Errors        1        0        0        0        0        0        0        0
                      Warnings        1        0        0        0        0        0        0        0
              Peer Connections        1        2        2        0        2        2        2        2
-- Inbound audio -----------------------------------------------------------------------------------
                          rate        2    28.73    14.36    14.36     0.00    28.73     0.00    28.73 Kbps
                          lost        1              0.00     0.00     0.00     0.00     0.00     0.00 %
                        jitter        2              0.00     0.00     0.00     0.00     0.00     0.00 s
          avgJitterBufferDelay        1             35.29     0.00    35.29    35.29    35.29    35.29 ms
-- Inbound video -----------------------------------------------------------------------------------
                      received        2     2.66     1.33     1.32     0.01     2.64     0.01     2.64 MB
                          rate        2   967.41   483.71   483.71     0.00   967.41     0.00   967.41 Kbps
                          lost        1              0.00     0.00     0.00     0.00     0.00     0.00 %
                        jitter        2              0.01     0.01     0.01     0.02     0.01     0.02 s
          avgJitterBufferDelay        1             50.48     0.00    50.48    50.48    50.48    50.48 ms
                         width        2               960      320      640     1280      640     1280 px
                        height        2               540      180      360      720      360      720 px
                           fps        1                15        0       15       15       15       15 fps
-- Outbound audio ----------------------------------------------------------------------------------
                          rate        2    42.84    21.42    21.42     0.00    42.84     0.00    42.84 Kbps
                          lost        1              0.00     0.00     0.00     0.00     0.00     0.00 %
                 roundTripTime        1             0.001    0.000    0.001    0.001    0.001    0.001 s
-- Outbound video ----------------------------------------------------------------------------------
                          sent        2     3.25     1.62     1.58     0.04     3.21     0.04     3.21 MB
                          rate        2  1131.25   565.63   565.63     0.00  1131.25     0.00  1131.25 Kbps
                          lost        1              0.00     0.00     0.00     0.00     0.00     0.00 %
                 roundTripTime        1             0.001    0.000    0.001    0.001    0.001    0.001 s
 qualityLimitResolutionChanges        2        2        1        1        0        2        0        2
          qualityLimitationCpu        2        0        0        0        0        0        0        0 %
    qualityLimitationBandwidth        2       20       10       10        0       20        0       20 %
           sentActiveEncodings        2                 2        1        1        3        1        3 encodings
                sentMaxBitrate        2  3700.00  1850.00   350.00  1500.00  2200.00  1500.00  2200.00 Kbps
                         width        2               640      640        0     1280        0     1280 px
                        height        2               360      360        0      720        0      720 px
                           fps        2                12       12        0       25        0       25 fps
              pliCountReceived        2                 1        0        1        2        1        2

Statistics values:

Name Count Description
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 %.
jitter Total streams The stream jitter in seconds.
avgJitterBufferDelay Total decoded tracks The inbound average jitter buffer delay.
qualityLimitResolutionChanges Total outbound video streams The qualityLimitationResolutionChanges value 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.

Prometheus / Grafana

See the prometheus stack.

Examples

Mediasoup demo

Starts one send-receive participant:

docker run -it --rm --name=webrtcperf-publisher \
    -v /dev/shm:/dev/shm \
    ghcr.io/vpalmisano/webrtcperf \
    --url=$MEDIASOUP_DEMO_URL \
    --url-query='roomId=test&displayName=Publisher($s-$t)' \
    --sessions=1 \
    --tabs-per-session=1

Starts 10 receive-only participants:

docker run -it --rm --name=webrtcperf-viewer \
    -v /dev/shm:/dev/shm \
    ghcr.io/vpalmisano/webrtcperf \
    --url=$MEDIASOUP_DEMO_URL \
    --url-query='roomId=test&displayName=Viewer($s-$t)&produce=false' \
    --sessions=1 \
    --tabs-per-session=10

Edumeet

Starts one send-receive participant, with a random audio activation pattern:

docker run -it --rm \
    -v /dev/shm:/dev/shm \
    -v $PWD/examples:/scripts:ro \
    ghcr.io/vpalmisano/webrtcperf \
    --url=$EDUMEET_URL \
    --url-query='displayName=Publisher($s-$t)' \
    --script-path=/scripts/edumeet-sendrecv.js \
    --sessions=1 \
    --tabs-per-session=1

Starts 10 receive-only participants:

docker run -it --rm \
    -v /dev/shm:/dev/shm \
    -v $PWD/examples:/scripts:ro \
    ghcr.io/vpalmisano/webrtcperf \
    --url=$EDUMEET_URL \
    --url-query='displayName=Viewer($s-$t)' \
    --script-path=/scripts/edumeet-recv.js \
    --sessions=1 \
    --tabs-per-session=10

Jitsi

Starts one send-receive participant:

docker run -it --rm \
    -v /dev/shm:/dev/shm \
    ghcr.io/vpalmisano/webrtcperf \
    --url=$JITSI_ROOM_URL \
    --url-query='#config.prejoinPageEnabled=false&userInfo.displayName=Participant($s-$t)' \
    --sessions=1 \
    --tabs-per-session=1

Starts 10 receive-only participants:

docker run -it --rm \
    -v /dev/shm:/dev/shm \
    ghcr.io/vpalmisano/webrtcperf \
    --url=$ROOM_URL \
    --url-query='#config.prejoinPageEnabled=false&userInfo.displayName=Participant($s-$t)' \
    --sessions=1 \
    --tabs-per-session=10

Running from source code

The DEBUG_LEVEL environment variable can be used to enable debug messages; see debug-level for syntax.

git clone https://github.com/vpalmisano/webrtcperf.git

cd webrtcperf

# Optional: 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_<version>-1_amd64.deb
# cd ..

yarn build

# sendrecv test
DEBUG_LEVEL=DEBUG:* yarn start \
    --url=https://127.0.0.1:3443/test \
    --url-query='displayName=SendRecv($s/$S-$t/$T)' \
    --script-path=./examples/edumeet-sendrecv.js \
    --sessions=1 \
    --tabs-per-session=1

# recv only
DEBUG_LEVEL=DEBUG:* yarn start \
    --url=https://127.0.0.1:3443/test \
    --url-query='displayName=Recv($s/$S-$t/$T)' \
    --script-path=./examples/edumeet-recv.js \
    --sessions=1 \
    --tabs-per-session=10

Using the VMAF calculator

  1. Run the tool adding the following options:
--script-params="{timestampWatermarkVideo:true,saveRecvVideoTrack:1}"
--server-port=5000
--server-use-https=true
--server-data=/data

With saveRecvVideoTrack you can specify the sessions that will be saved at receiver side (in this example 1 will save all the streams received in the sessions with index 0 and 1). 2. The sent/received videos will be saved in the /data directory. 3. Run the VMAF calculator comparing the sent/received videos:

docker run --rm \
  -e DEBUG_LEVEL=INFO \
  -v $PWD/data:/data \
  ghcr.io/vpalmisano/webrtcperf:devel \
  --vmaf-path /data \

The tool will generate a .vmaf.json and a .plotly files in the data/vmaf directory. Adding the --vmaf-preview option, a .mp4 file containing the side-by-side video comparison will be generated.

Authors

License

AGPL

Open Source Agenda is not affiliated with "Webrtc Stress Test" Project. README Source: vpalmisano/webrtcperf

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