Hello Kamailio Community,

I recently encountered an interesting behavior when using Kamailio to relay SIP messages from a WebSocket client to FreeSWITCH over UDP.

Observed Behavior:

• The WebSocket client sends an INVITE with a large SDP.
• Kamailio forwards this request over UDP to FreeSWITCH.
• Upon inspection, the SDP in the relayed message appears truncated. (Probably MTU limit)
• Surprisingly, the call still establishes successfully, and there are no noticeable issues with audio or call setup.

My Questions:

1. Is this expected behavior? Should Kamailio automatically truncate SDP when relaying from WebSocket to UDP?
2. Could this be working accidentally? For example, is FreeSWITCH handling the partial SDP gracefully by default?
3. Should I be concerned about potential failures in different scenarios? (e.g., ICE candidate loss, missing codec negotiation)

I would appreciate any insights from the community on whether this is a known or expected behavior in Kamailio, or if it might be a configuration issue.

Thanks in advance for your help!

Best regards,
Pavan Kumar