sip-to-webrtc

sip-to-webrtc demonstrates how you can connect to a SIP over WebRTC endpoint. This example connects to an extension and saves the audio to a ogg file.

Instructions

Setup FreeSWITCH (or SIP over WebSocket Server)

With a fresh install of FreeSWITCH all you need to do is

• Enable ws-binding
• Set a default_password to something you know

Run sip-to-webrtc

Run go run *.go -h to see the arguments of the program. If everything is working this is the output you will see.

$ go run *.go -host 172.17.0.2 -password Aelo1ievoh2oopooTh2paijaeNaidiek
  Connection State has changed checking
  Connection State has changed connected
  Got Opus track, saving to disk as output.ogg
  Connection State has changed disconnected

Play the audio file

ffmpeg's in-tree Opus decoder isn't able to play the default audio file from FreeSWITCH. Use the following command to force libopus.

ffplay -acodec libopus output.ogg