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-users_at_lists.kamailio.org_airsay@duck.com 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! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
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 sr-users@lists.kamailio.org Sent: Mittwoch, 23. April 2025 19:14 To: sr-users@lists.kamailio.org Cc: Fernando Lopes fernandolopes20003@gmail.com; sr- users@lists.kamailio.org; airsay@duck.com 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-
users_at_lists.kamailio.org_airsay@duck.com> 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! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr- users@lists.kamailio.org To unsubscribe send an email to sr-users- leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
Try using network=host instead of node port
Regards,
David Villasmil email: david.villasmil.work@gmail.com
On Thu, Apr 24, 2025 at 7:42 PM Henning Westerholt via sr-users < sr-users@lists.kamailio.org> 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 sr-users@lists.kamailio.org Sent: Mittwoch, 23. April 2025 19:14 To: sr-users@lists.kamailio.org Cc: Fernando Lopes fernandolopes20003@gmail.com; sr- users@lists.kamailio.org; airsay@duck.com 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-
users_at_lists.kamailio.org_airsay@duck.com> 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! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only
to the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr- users@lists.kamailio.org To unsubscribe send an email to sr-users- leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to
the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
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 < sr-users@lists.kamailio.org> 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" sr-users@lists.kamailio.org *An: *"Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org *CC: *"Fernando Lopes" fernandolopes20003@gmail.com,"airsay@duck.com" < airsay@duck.com>,"David Villasmil" david.villasmil.work@gmail.com *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: david.villasmil.work@gmail.com
On Thu, Apr 24, 2025 at 7:42 PM Henning Westerholt via sr-users < sr-users@lists.kamailio.org> 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 sr-users@lists.kamailio.org Sent: Mittwoch, 23. April 2025 19:14 To: sr-users@lists.kamailio.org Cc: Fernando Lopes fernandolopes20003@gmail.com; sr- users@lists.kamailio.org; airsay@duck.com 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-
users_at_lists.kamailio.org_airsay@duck.com> 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! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only
to the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr- users@lists.kamailio.org To unsubscribe send an email to sr-users- leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only
to the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
__________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
You should use network=host, exposing each port will take forever to start and it’s not optimal.
Regards,
David Villasmil email: david.villasmil.work@gmail.com
On Mon, Apr 28, 2025 at 11:36 AM Duncan Turnbull via sr-users < sr-users@lists.kamailio.org> 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 < sr-users@lists.kamailio.org> 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" sr-users@lists.kamailio.org *An: *"Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org *CC: *"Fernando Lopes" fernandolopes20003@gmail.com,"airsay@duck.com" < airsay@duck.com>,"David Villasmil" david.villasmil.work@gmail.com *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: david.villasmil.work@gmail.com
On Thu, Apr 24, 2025 at 7:42 PM Henning Westerholt via sr-users < sr-users@lists.kamailio.org> 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 sr-users@lists.kamailio.org Sent: Mittwoch, 23. April 2025 19:14 To: sr-users@lists.kamailio.org Cc: Fernando Lopes fernandolopes20003@gmail.com; sr- users@lists.kamailio.org; airsay@duck.com 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-
users_at_lists.kamailio.org_airsay@duck.com> 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! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply
only to the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr- users@lists.kamailio.org To unsubscribe send an email to sr-users- leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only
to the
sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
__________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
I'm meaning to reply to OP, but not sure how to do that yet. I had a similar issue with a slow/buggy video doorbell that spoke SIP.
In pjsip.conf I changed timer_b=32000 to 120000, the maximum length of call that doorbell can even make. Solved my calls being cut off at 32 seconds.
After a lot of messages I'm trying to resolve the problem. Problem: Kamailio can't respond ACK to asterisk because wrong contact is being sent by asterisk
-
airsay＠duck.com -
David Villasmil -
Duncan Turnbull -
Fernando Lopes -
Hans-Jürgen Brand -
Henning Westerholt -
joshruehlig＠gmail.com