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Kamailio fails to start because of...

Jean Cérien

28 Mar 2018 28 Mar '18

12:48 a.m.

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.

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Alberto Llamas

28 Mar 28 Mar

1:06 a.m.

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(a)gmail.com> wrote:

...

-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA *"Internet is all about share"*

Mack Hendricks

3:04 p.m.

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(a)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(a)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(a)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(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Jean Cérien

4:04 p.m.

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(a)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(a)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(a)gmail.com> wrote:

-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA *"Internet is all about share"* _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Mack Hendricks

4:20 p.m.

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(a)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(a)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(a)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(a)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(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

_______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Jean Cérien

29 Mar 29 Mar

1:09 a.m.

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(a)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(a)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(a)goflyball.com> wrote:

Joel Serrano

1:25 a.m.

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(a)gmail.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(a)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(a)gmail.com> wrote:

-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA "Internet is all about share" _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

_______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Jean Cérien

10:43 p.m.

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(a)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(a)gmail.com> wrote:

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(a)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(a)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(a)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(a)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(a)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

> 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(a)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(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

_______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Fred Posner

28 Mar 28 Mar

1:06 a.m.

On 3/27/18 5:48 PM, Jean Cérien wrote:

...

ESX/VMware has been great for me with kamailio, even with RTP and high volume. --fred

Jean Cérien

1:21 a.m.

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(a)palner.com> wrote:

...

On 3/27/18 5:48 PM, Jean Cérien wrote:

volume. --fred _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Daniel Tryba

11:29 a.m.

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.

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participants (7)

Alberto Llamas
Daniel Tryba
Fred Posner
Jean Cérien
Joel Serrano
Mack Hendricks
Mack Hendricks