webrtc-sip-client

WebRTC SIP client on golang for FreeSwitch

WebRTC SIP client for imitate webrtc client from browser.

Tested only with FreeSwitch 1.10 webrtc server. Codec OPUS with 8000hz bandwith.

Use cases

• Integration/functional tests webrtc server in CICD
• Stress test stage/production webrtc servers
• As client in development process

Usage

Send invite to wss://webrtc.site.com/webrtc with concurency 10

go run main.go --host webrtc.site.com --invite 0000 --transport wss --port 443 --path /webrtc -c 10

Connect to wss://webrtc.site.com/webrtc and wait invite from webrtc server

go run main.go --host webrtc.site.com --transport wss --port 443 --path /webrtc -c 1

Arguments

Usage: main [--count COUNT] [--invite NUMBER] [--username USERNAME] [--password PASSWORD] [--domain DOMAIN] [--transport TRANSPORT] [--host HOST] [--path PATH] [--port PORT] [--savetofile] [--outfilename FILENAME] [--infilename FILENAME] [--srtpkey PATH] [--srtpcert PATH] [--progress] [--verbose]

Options:
  --count COUNT, -c COUNT
                         Count instances [default: 1]
  --invite NUMBER, -i NUMBER
                         Number for invite
  --username USERNAME [default: 101]
  --password PASSWORD [default: 101]
  --domain DOMAIN [default: local]
  --transport TRANSPORT [default: ws]
  --host HOST [default: 192.168.100.10]
  --path PATH            Path in server, for examples /webrtc/socket
  --port PORT [default: 5071]
  --savetofile, -s       Save media to file in ogg format --outfilename [default: false]
  --outfilename FILENAME [default: output.ogg]
  --infilename FILENAME
                         Play ogg file in channel, example: --infilename input.ogg
  --srtpkey PATH [default: certs/dtls-srtp.pem]
  --srtpcert PATH [default: certs/dtls-srtp.pub.pem]
  --progress, -p         Display rtp progress [default: false]
  --verbose, -v          Verbose [default: false]
  --help, -h             display this help and exit