PBX — WebRTC Softphone and Browser Calling

Overview

The SipLive softphone enables calls from any modern web browser using WebRTC. No software install needed. Signaling uses SIP over WebSocket (WSS); media uses SRTP.

How It Works

1. Browser connects to wss://<pbx-host>:8089/ws.
2. Authenticates using the extension's -wss endpoint credentials.
3. Incoming calls arrive as SIP INVITE over WebSocket; browser handles audio via WebRTC API.
4. Media flows as SRTP between browser and PBX.

Requirements

• Valid CA-signed TLS certificate — Browsers reject self-signed certs for WSS.
• SRTP enabled on the WSS endpoint — WebRTC mandates encrypted media.
• Separate -wss endpoint and auth — Sharing auth between UDP and WSS causes silent failures.
• Port 8089/TCP open in the firewall.
• Modern browser — Chrome 80+, Firefox 78+, Edge 80+, Safari 14+. Microphone permission required.

WSS Endpoint Settings (ps_endpoints)

• transport = transport-wss
• media_encryption = sdes
• webrtc = yes
• dtls_verify = no

These are set automatically when creating a WebRTC extension via the admin portal.

Opening the Softphone

Users click the Phone button in the portal nav bar. It opens in a popup, auto-registers, and stays active while open. Browser notifications alert for incoming calls.

Troubleshooting

• Connection refused on 8089 — Check asterisk.conf [http] has enabled=yes and firewall allows TCP 8089.
• Registered but no audio — Verify RTP ports 10000–20000 are open and SRTP is negotiating.
• 401 on WSS — The -wss endpoint is missing its own ps_auths row.
• One-way audio — NAT issue. Set external_media_address and external_signaling_address to the PBX's public IP.