webrtc-load-tool

Tool for simulating WebRTC viewers and load testing WebRTC streams

webrtc-load-tool v1.0.0

Usage

webrtc-load-tool WHIP-URL [flags]

Flags

• -b, --bufferduration duration Buffer duration for RTP jitter buffer for lost packets counter (default 500ms)
• -c, --connections string Maximum number of connections to create (default: "1")
• --codec string Preferred codec (H264 or VP8). Defaults to H264 if resolution is set
• -d, --duration duration Time to run the test (default: 1m0s)
• -l, --lite Lite mode, no RTP video/audio parsing
• -m, --relaymode string Relay mode to use (auto, no, only) (default: "auto")
• -r, --runup duration Time frame to create the maximum number of connections
• -t, --targetresolution string Target resolution height. Automatically switches to the closest available resolution <= target. (websocket mode only) Accepts: 4k, 2160p, 2160, 2k, 1440p, 1440, 1080p, 1080, hd, fhd, 720p, 720, hdready, 480p, 480, 360p, 360, 240p, 240, or a numeric value. Set to empty to disable resolution switching.

Example

webrtc-load-tool https://fly.live.ceeblue.tv/webrtc/out%20de1e6f7c-e5db-450b-9603-c3644274779b?video=3 -c 10 -r 10s -d 1m

target resoultion (Ceeblue websocket protocol):

webrtc-load-tool wss://fly.live.ceeblue.tv/webrtc/out+de1e6f7c-e5db-450b-9603-c3644274779b -c 1 -r 10s -d 1m -t 720p