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

You should use network=host, exposing each port will take forever to start
and it’s not optimal.


Regards,

David Villasmil
email: [email protected]



On Mon, Apr 28, 2025 at 11:36 AM Duncan Turnbull via sr-users <
[email protected]> wrote:

> One thing I find is helpful is if you can tcpdump or sngrep or wireshark
> on one or both of the devices to see the a SIP packet debug. Wireshark has
> a good voip analyser and takes a pcap from tcpdump which is easier to read.
> sngrep is a great text tool for sip.
>
> The RTP response to the ACK maybe being sent to the wrong place or can't
> get to where it needs to go. You will see where the ACK and what addresses
> are being used. Often a NAT firewall may be blocking itor it
>
>
>
>
> On Mon, 28 Apr 2025 at 19:59, Hans-Jürgen Brand via sr-users <
> [email protected]> wrote:
>
>>
>>
>> > > Do I need to explicitly expose the RTP port range (10000–20000) in the
>> > Asterisk Kubernetes service as well
>>
>> For my understandig of kubernetes yes. otherwise you can not reach the
>> udp outside the kubernetes cluster
>>
>> kind regards
>> Hans-Jürgen
>>
>> *Gesendet: *Freitag, 25. April 2025 um 03:27
>> *Von: *"David Villasmil via sr-users" <[email protected]>
>> *An: *"Kamailio (SER) - Users Mailing List" <[email protected]>
>> *CC: *"Fernando Lopes" <[email protected]>,"[email protected]" <
>> [email protected]>,"David Villasmil" <[email protected]>
>> *Betreff: *[SR-Users] Re: SIP Call Drops After 0.32 Seconds – Asterisk
>> 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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