Hello everyone, I'm running into an issue with SIP calls in my current setup and would really appreciate some help.
Setup: I have a machine named sip00 (IP: 192.168.1.75) running Kamailio + RTPengine. Kamailio is dispatching calls to sip:192.168.1.190:32210;transport=tcp. This IP points to another machine running Asterisk inside a Kubernetes cluster. RTPengine is configured with an RTP port range of 10000–20000, and my router is set to allow that same range.
Asterisk Kubernetes Service Configuration: yaml Copy Edit spec: ports: - name: tcp-port protocol: TCP port: 5060 targetPort: 5060 nodePort: 32210 - name: udp-port protocol: UDP port: 5060 targetPort: 5060 nodePort: 32210
Problem: When I initiate a SIP call, the router forwards traffic to Kamailio + RTPengine, which then sends it to the Asterisk server on 192.168.1.190. Everything seems fine initially, but at exactly 0.32 seconds into the call, Asterisk sends a BYE and no longer responds with 200 OK to the SIP dialog — even though I'm still receiving and sending audio. Then, at around 01:04, I get a 408 Request Timeout.
Questions: Do I need to explicitly expose the RTP port range (10000–20000) in the Asterisk Kubernetes service as well? Why is Asterisk sending a BYE so early if audio is still flowing? Could it be a signaling timeout or an issue with SIP dialog tracking? Any help or pointers would be greatly appreciated!
Thanks in advance!