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

Try using network=host instead of node port

Regards,

David Villasmil
email: [email protected]



On Thu, Apr 24, 2025 at 7:42 PM Henning Westerholt via sr-users <
[email protected]> wrote:

> Hello,
>
> Does the BYE comes after 320ms or 32s? As said, look to the asterisk log
> why it sends the BYE, maybe it don't receive media.
> Can the rtpengine actually expose ports to the internal and external
> network? Do you see RTP traffic flowing between the different services?
>
> Cheers,
>
> Henning
>
> > -----Original Message-----
> > From: airsay--- via sr-users <[email protected]>
> > Sent: Mittwoch, 23. April 2025 19:14
> > To: [email protected]
> > Cc: Fernando Lopes <[email protected]>; sr-
> > [email protected]; [email protected]
> > Subject: [SR-Users] Re: SIP Call Drops After 0.32 Seconds – Asterisk 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 <sr-
> > [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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> >
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