webrtc-bench is a WebRTC server that plays a video file from disk, and then re-distributes via WebRTC. It generates performance metrics including CPU usage, peer connection counts, and detailed latency measurements for connection setup phases.

This repo contains a CLI WebRTC Client that will connect and perform the load test. This is done via a REST API hosted on :8080/doSignaling. You can also watch the video/load test via the web UI at :8080/, it connects via the same REST API as the CLI.

• git clone https://github.com/pion/webrtc-bench.git
• cd webrtc-bench/server
• ffmpeg -i $INPUT_FILE -g 30 -b:v 2M input.ivf
• go run main.go

You can then access the Web UI at https://localhost:8080. This requires a browser that supports H264, this repo doesn't do any error handling around signaling failures.

You can also use the provided WebRTC CLI

• cd webrtc-bench/client
• go run main.go localhost:8080

The benchmark generates three CSV files with performance data:

System-level metrics collected every 3 seconds:

timestamp, peerConnectionCount, cpuUsage, droppedLatencyMeasurements
2020-12-15T13:59:23-08:00, 1, 17.878028, 0
2020-12-15T13:59:26-08:00, 0, 35.101254, 0
2020-12-15T13:59:29-08:00, 1, 12.565718, 0

Detailed WebRTC connection setup timing from server perspective:

timestamp,type,latencyMs
2026-01-02T19:33:57.714Z,signaling_processing,13.391
2026-01-02T19:33:57.714Z,sdp_offer_processing,6.587
2026-01-02T19:33:57.714Z,sdp_answer_creation,0.365
2026-01-02T19:33:57.715Z,ice_gathering,0.255
2026-01-02T19:33:57.825Z,ice_connection,122.588
2026-01-02T19:33:57.979Z,dtls_handshake,153.612

Server Metrics:

• signaling_processing: Total signaling time (offer → answer)
• sdp_offer_processing: SDP offer parsing time
• sdp_answer_creation: SDP answer generation time
• ice_gathering: ICE candidate gathering time
• ice_connection: Time from connection start to ICE connected
• dtls_handshake: DTLS handshake completion time

Connection setup timing from client perspective:

timestamp,type,latencyMs
2026-01-02T11:33:57.706Z,ice_gathering,5001.234
2026-01-02T11:33:57.706Z,signaling_rtt,72.456
2026-01-02T11:33:57.811Z,ice_connection,5183.789
2026-01-02T11:33:57.916Z,dtls_handshake,104.123
2026-01-02T11:33:57.916Z,media_ready,5287.912

Client Metrics:

• ice_gathering: ICE candidate gathering time
• signaling_rtt: Round-trip time for signaling exchange
• ice_connection: Total time to ICE connected
• dtls_handshake: DTLS handshake completion time
• media_ready: Total connection setup time

All latency measurements are recorded with sub-millisecond precision.

The following graph was generated from the results.csv that was generated by the server. The client was run from another EC2 instance in the same VPC. The Y Axis is the CPU Usage, and the X Axis is the count of PeerConnections. This was run on a m4.2xlarge (Intel(R) Xeon(R) CPU E5-2686 v4 @ 2.30GHz)

This is controlled by what video you use. You can go as high (or low) as you want.

SRTP_AEAD_AES_128_GCM

None, on purpose.

Pion by design allows the user to specify what they want. We are distributing a video feed that is already encoded, and has a very small keyframe-interval so these things don't help us.

When setting up the PeerConnection you have full control of the RTP/RTCP pipelines, so users can add TWCC/NACK/FEC in a way that works for them.

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If you need commercial support or don't want to use public methods you can contact us at team@pion.ly

MIT License - see LICENSE for full text