Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Hi Jean,
It might be something else. We do have an entire virtualized environment on Vmware with Asterisk, kamailios and another VoIP component without any issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com http://dopensource.com/ support: +888-907-2085 dSIPRouter http://dsiprouter.org/ - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas <albertollamaso@gmail.com mailto:albertollamaso@gmail.com> wrote:
Hi Jean,
It might be something else. We do have an entire virtualized environment on Vmware with Asterisk, kamailios and another VoIP component without any issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien <cerien.jean@gmail.com mailto:cerien.jean@gmail.com> wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org mailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
"Internet is all about share" _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org mailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio.
Here is the situation. Call arrives from voip provider to kamailio, it dispatches to asterisk, asterisk answers, and initiates another call through kamailio, and the voip provider.
K <-----------> Asterisk Invite -> <--- 100 <----180 <--- 200 <--- 200 retransmission,; happens 3-5 times Invite --> (same callid & cseq) <--- 200 retransmission,; happens 3-5 times
So, we see the asterisk dialplan has answered, and another call is placed form the asterisk K <-----------> Asterisk <------Invite 100 ----> (2 or 4 seconds later) 408 ---->
both nodes (kamailio and asterisk) show the same traces.
Any ideas would be greatly & truly appreciated, I am getting quite desperate about this one !
J.
On Wed, Mar 28, 2018 at 8:04 AM, Mack Hendricks ap@goflyball.com wrote:
Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
*Mack Hendricks / Head of Support / dOpenSource* web: http://dopensource.com support: +888-907-2085 dSIPRouter http://dsiprouter.org/ - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Jean,
It might be something else. We do have an entire virtualized environment on Vmware with Asterisk, kamailios and another VoIP component without any issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
*"Internet is all about share"* _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Is the 200 getting back to the carrier? I’m assuming not. What does the INVITE and 200 message look like
On Mar 28, 2018, at 9:04 AM, Jean Cérien cerien.jean@gmail.com wrote:
Kamailio.
Here is the situation. Call arrives from voip provider to kamailio, it dispatches to asterisk, asterisk answers, and initiates another call through kamailio, and the voip provider.
K <-----------> Asterisk Invite -> <--- 100 <----180 <--- 200 <--- 200 retransmission,; happens 3-5 times Invite --> (same callid & cseq) <--- 200 retransmission,; happens 3-5 times
So, we see the asterisk dialplan has answered, and another call is placed form the asterisk K <-----------> Asterisk <------Invite 100 ----> (2 or 4 seconds later) 408 ---->
both nodes (kamailio and asterisk) show the same traces.
Any ideas would be greatly & truly appreciated, I am getting quite desperate about this one !
J.
On Wed, Mar 28, 2018 at 8:04 AM, Mack Hendricks ap@goflyball.com wrote: Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com support: +888-907-2085 dSIPRouter - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Jean,
It might be something else. We do have an entire virtualized environment on Vmware with Asterisk, kamailios and another VoIP component without any issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
"Internet is all about share" _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Thanks for the help
I've reproduced the issue on the test bed, with sipp to generate calls.
The issue appears in the second call - Asterisk places a call to Kamailio that should relay it to the carrier. Asterisk sends Invite, Kamailio replies with 100 and then nothing gets out of kamailio (I use sngrep on the box). I have traces in various routes in K, I see the call to t_relay, but I see nothing in sngrep - 2 or 4 secs later, K generates the 408
J.
On Wed, Mar 28, 2018 at 9:20 AM, Mack Hendricks mack@dopensource.com wrote:
Is the 200 getting back to the carrier? I’m assuming not. What does the INVITE and 200 message look like
On Mar 28, 2018, at 9:04 AM, Jean Cérien cerien.jean@gmail.com wrote:
Kamailio.
Here is the situation. Call arrives from voip provider to kamailio, it dispatches to asterisk, asterisk answers, and initiates another call through kamailio, and the voip provider.
K <-----------> Asterisk Invite -> <--- 100 <----180 <--- 200 <--- 200 retransmission,; happens 3-5 times Invite --> (same callid & cseq) <--- 200 retransmission,; happens 3-5 times
So, we see the asterisk dialplan has answered, and another call is placed form the asterisk K <-----------> Asterisk <------Invite 100 ----> (2 or 4 seconds later) 408 ---->
both nodes (kamailio and asterisk) show the same traces.
Any ideas would be greatly & truly appreciated, I am getting quite desperate about this one !
J.
On Wed, Mar 28, 2018 at 8:04 AM, Mack Hendricks ap@goflyball.com wrote:
Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
*Mack Hendricks / Head of Support / dOpenSource* web: http://dopensource.com support: +888-907-2085 dSIPRouter http://dsiprouter.org/ - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Jean,
It might be something else. We do have an entire virtualized environment on Vmware with Asterisk, kamailios and another VoIP component without any issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
*"Internet is all about share"* _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Where are you t_relay()ing to?
Do you use dispatcher or similar?
Are you manually setting $du?
On Wed, Mar 28, 2018 at 3:09 PM, Jean Cérien cerien.jean@gmail.com wrote:
Thanks for the help
I've reproduced the issue on the test bed, with sipp to generate calls.
The issue appears in the second call - Asterisk places a call to Kamailio that should relay it to the carrier. Asterisk sends Invite, Kamailio replies with 100 and then nothing gets out of kamailio (I use sngrep on the box). I have traces in various routes in K, I see the call to t_relay, but I see nothing in sngrep - 2 or 4 secs later, K generates the 408
J.
On Wed, Mar 28, 2018 at 9:20 AM, Mack Hendricks mack@dopensource.com wrote:
Is the 200 getting back to the carrier? I’m assuming not. What does the INVITE and 200 message look like
On Mar 28, 2018, at 9:04 AM, Jean Cérien cerien.jean@gmail.com wrote:
Kamailio.
Here is the situation. Call arrives from voip provider to kamailio, it dispatches to asterisk, asterisk answers, and initiates another call through kamailio, and the voip provider.
K <-----------> Asterisk Invite -> <--- 100 <----180 <--- 200 <--- 200 retransmission,; happens 3-5 times Invite --> (same callid & cseq) <--- 200 retransmission,; happens 3-5 times
So, we see the asterisk dialplan has answered, and another call is placed form the asterisk K <-----------> Asterisk <------Invite 100 ----> (2 or 4 seconds later) 408 ---->
both nodes (kamailio and asterisk) show the same traces.
Any ideas would be greatly & truly appreciated, I am getting quite desperate about this one !
J.
On Wed, Mar 28, 2018 at 8:04 AM, Mack Hendricks ap@goflyball.com wrote:
Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com support: +888-907-2085 dSIPRouter - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Jean,
It might be something else. We do have an entire virtualized environment on Vmware with Asterisk, kamailios and another VoIP component without any issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
"Internet is all about share" _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Many thanks for the help, I think I am getting closer, and obviously ;-) the issue is not caused by Kamailio !
under some load (still to confirm); when asterisk receives the relayed invite from K, it generates the reply message (100 trying) but this message does not reach the network, ie, I see the message in asterisk debug log, but not in the sngrep/pcap traces
Again, thanks for the help !
J.
On Wed, Mar 28, 2018 at 6:25 PM, Joel Serrano joel@gogii.net wrote:
Where are you t_relay()ing to?
Do you use dispatcher or similar?
Are you manually setting $du?
On Wed, Mar 28, 2018 at 3:09 PM, Jean Cérien cerien.jean@gmail.com wrote:
Thanks for the help
I've reproduced the issue on the test bed, with sipp to generate calls.
The issue appears in the second call - Asterisk places a call to Kamailio that should relay it to the carrier. Asterisk sends Invite, Kamailio replies with 100 and then nothing gets
out
of kamailio (I use sngrep on the box). I have traces in various routes in K, I see the call to t_relay, but I
see
nothing in sngrep - 2 or 4 secs later, K generates the 408
J.
On Wed, Mar 28, 2018 at 9:20 AM, Mack Hendricks mack@dopensource.com wrote:
Is the 200 getting back to the carrier? I’m assuming not. What does
the
INVITE and 200 message look like
On Mar 28, 2018, at 9:04 AM, Jean Cérien cerien.jean@gmail.com wrote:
Kamailio.
Here is the situation. Call arrives from voip provider to kamailio, it dispatches to asterisk, asterisk answers, and initiates another call
through
kamailio, and the voip provider.
K <-----------> Asterisk Invite -> <--- 100 <----180 <--- 200 <--- 200 retransmission,; happens 3-5 times Invite --> (same callid & cseq) <--- 200 retransmission,; happens 3-5 times
So, we see the asterisk dialplan has answered, and another call is
placed
form the asterisk K <-----------> Asterisk <------Invite 100 ----> (2 or 4 seconds later) 408 ---->
both nodes (kamailio and asterisk) show the same traces.
Any ideas would be greatly & truly appreciated, I am getting quite desperate about this one !
J.
On Wed, Mar 28, 2018 at 8:04 AM, Mack Hendricks ap@goflyball.com
wrote:
Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com support: +888-907-2085 dSIPRouter - GUI focused on implementing Kamailio to provide SIP
Trunking
and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Jean,
It might be something else. We do have an entire virtualized
environment
on Vmware with Asterisk, kamailios and another VoIP component without
any
issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I
actually
have no audio issues, but communication between the asterisk &
kamailio for
sip sometime fails - I get a few 408. I cant tell if this is network
related
or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
"Internet is all about share" _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On 3/27/18 5:48 PM, Jean Cérien wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
ESX/VMware has been great for me with kamailio, even with RTP and high volume.
--fred
Many thanks for this quick feedback. Is that your own hardware, or something hosted ?
J.
On Tue, Mar 27, 2018 at 6:06 PM, Fred Posner fred@palner.com wrote:
On 3/27/18 5:48 PM, Jean Cérien wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
ESX/VMware has been great for me with kamailio, even with RTP and high
volume.
--fred
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On Tue, Mar 27, 2018 at 06:06:58PM -0400, Fred Posner wrote:
Anyone has advice on kamailio on a VM, when it only handles sip?? ?
ESX/VMware has been great for me with kamailio, even with RTP and high volume.
ESX works fine indeed. But when you run something like rtpengine with a moderate volume in a guest, from the guest its perspective all if well (near 0% CPU usage). From the host its perspective it spends a large amount of CPU time just scheduling the network interrupts for all those tiny RTP packets, the amount is largely dependent on the ethernet cards.
-
Alberto Llamas -
Daniel Tryba -
Fred Posner -
Jean Cérien -
Joel Serrano -
Mack Hendricks -
Mack Hendricks