🚀 New Desktop Feature: Built-In Proxy for Local PBX Connections

[InnovateAsterisk]

We are dying to know if this simple (but very cool) patch is working for you guys!

We’ve just introduced a built-in proxy inside the Siperb Desktop App — and it solves a long-standing integration problem for local PBX deployments.

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🔎 The Previous Problem

Siperb Softphone connects beautifully to your local PBX when:

• Registration Mode = WebSocket
• Your PBX (Asterisk / FreePBX / etc.) supports WebSockets (WS/WSS)

But what if your PBX:

• ❌ Does not support WebSockets
• ❌ Only exposes UDP, TCP, or TLS SIP ports
• ❌ Is an older or more locked-down deployment

Until now… that was a blocker.

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✅ The Solution: Built-In Proxy Mode

You can now:

1. Select WebSocket Registration Mode
2. Choose your Transport:
   • UDP
   • TCP
   • TLS
   • (or WSS if you want direct connection)

If you choose UDP, TCP, or TLS, the Desktop App will:

• Start a lightweight local WebSocket server
• Connect the softphone back to itself
• Proxy SIP signalling to your PBX using the selected transport

👉 The result?
You still get full WebRTC behaviour, even if your PBX doesn’t support WebSockets.

Yes. Still WebRTC. Still modern. Still Siperb.

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🔐 Security Notes (Important)

• UDP and TCP are insecure transports
• The SIP INVITE will still include the DTLS fingerprint in the SDP
• This mode is intended for local PBX environments only

We believe this is acceptable for LAN deployments — but we do not recommend using UDP/TCP on public networks.

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🖥 Firewall / Port Reminder

Because the app starts a local WebSocket server:

• The configured port must be open on your PC
• Or the app must be whitelisted in your firewall

If the port is blocked, the softphone won’t be able to connect back to itself.

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🎙 About Media (Read Carefully)

Signalling is handled via the new proxy.

But media?

Media is still transmitted:

• Directly from the browser
• Using DTLS-SRTP
• Securely
• Peer-to-server

This means your PBX must accept WebRTC media.

For Asterisk, you can simply enable:

webrtc=yes

Media negotiation still requires proper WebRTC support on the server side.

We are working on improving media flexibility — but signalling was the first major hurdle to remove.

[InnovateAsterisk]