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