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
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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!
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