A minimal demo of SIP ↔ WebRTC. This example shows how a WebRTC browser client can register, call, and communicate through a SIP signaling server.

Built using Pion and Emiago SIP.

Quick Start

# Clone and start
git clone https://github.com/arjunshajitech/sip-webrtc-gateway.git

# Go to webrtc-sip folder or sip-webrtc folder
cd webrtc-sip
# or
cd sip-webrtc

# Run
go run .

you should see:

webrtc server initialized, http://localhost:5000
sip server initialized, 192.168.1.37:5060/udp

Note: 192.168.1.37 is your IP address - it will show your actual IP

Testing

1. Load the page at http://localhost:5000 in Chrome
2. Click "Start"
3. Use a Polycom to dial 192.168.1.37 (your IP address)
4. Check logs - you should see:

   accepting sip invite 
   

5. Call is now connected!

Known Issues

• Video may stuck because keyframe handling is not implemented
• May see some panic errors if hardcoded UDP ports are busy on your system - just ignore or free the ports
• Sometimes calling via Polycom fails because the app is not stable as of now - be sure to wait for "accepting sip invite" in logs before attempting another call