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