[BUG / QUESTION] WebRTC → UDP via dSIP: Contact uses TLS/WSS instead of UDP?

[Antonio]

Hi all,
just want to confirm if I understand this correctly.My goal is to use dSIPRouter as a proxy to convert WebRTC SIP (WSS) to SIP over UDP, while keeping media direct between the WebRTC client and FreePBX.I’ve configured the FreePBX PJSIP endpoint like in the attached screenshot.With this setup:

• the WebRTC extension registers via dSIP,
• I can reach the PBX,
• calls work and RTP is sent directly (no media relay through dSIP) — so far so good.

However, the REGISTER that arrives on the PBX from dSIP has a Contact like:

```
Contact: "226"<sips:2...@domain.name;rtcweb-breaker=yes;transport=wss;domain=domain.name;exten=226>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
```

Because of this, the PBX sends OPTIONS to domain.name over TLS/WSS.I was not  expecting that, since dSIP is converting WebRTC to UDP, the PBX would instead be sending OPTIONS over UDP (i.e. Contact pointing to the dSIP UDP address), not over TLS/WSS.Am I missing something in the configuration, or is this a bug in how dSIP is rewriting the Contact/transport?

I tested on 2 different dSIP (with Extennal IP and behind NAT)