Re: SIP Call Drops After 0.32 Seconds – Ast erisk on Kubernetes with Kamailio + RTPengine

I will let the more experienced Kamailio folks comment on the technical/Kamailio/RTP modifications that you need to make. The SIP standard requires an ACK to a 200 OK to be sent within that 32 second window. If an ACK is not received, the session is torn down. You would need to do a packet capture to review the Contact header in the SIP and the c: in the SDP. I'm just getting bedded in with SIP, reading Alan B Johnston's "SIP: Understanding the Session Initiation Protocol". Excellent introduction to understanding SIP.

Best regards 
Sent from my iPhone

> On 23 Apr 2025, at 16:35, Fernando Lopes via sr-users <[email protected]> wrote:
> 
> 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!
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