Hi List,
I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing.
So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios.
For testing purposes I setup an freeswitch with beep and then echo application to the caller.
My call flow are like this
Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together.
This generates for the caller an beep and echo.
The interesting thing is now, that Call 1 gets an hearable delay of 1 second.
But only in the rtp steam from me to the caller.
The second calls seems equal of timing.
Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no.
Now my question to you guys.
How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc).
Only capturing on one place don't did the trick for me.
Thanks for your hints
Cheers Karsten
Kamailio does not handle RTP.
But you can use heplify ( or homer) That will alliw you to collect rtcp in your infrastructure and map it with the SIP calls you have.
On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, khorsmann@gmail.com wrote:
Hi List,
I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing.
So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios.
For testing purposes I setup an freeswitch with beep and then echo application to the caller.
My call flow are like this
Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together.
This generates for the caller an beep and echo.
The interesting thing is now, that Call 1 gets an hearable delay of 1 second.
But only in the rtp steam from me to the caller.
The second calls seems equal of timing.
Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no.
Now my question to you guys.
How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc).
Only capturing on one place don't did the trick for me.
Thanks for your hints
Cheers Karsten _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi yuriy,
Of course per default Kamailio don't handle rtp. I forgot that I use rtpengine on the "sbc" with Kamailio.
And as you all know, if it's sounds bad, then we sip guys get the tickets...
So that's an question for your experience in this area.
Cheers Karsten
Yuriy Gorlichenko ovoshlook@gmail.com schrieb am Do., 6. Feb. 2020, 10:27:
Kamailio does not handle RTP.
But you can use heplify ( or homer) That will alliw you to collect rtcp in your infrastructure and map it with the SIP calls you have.
On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, khorsmann@gmail.com wrote:
Hi List,
I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing.
So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios.
For testing purposes I setup an freeswitch with beep and then echo application to the caller.
My call flow are like this
Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together.
This generates for the caller an beep and echo.
The interesting thing is now, that Call 1 gets an hearable delay of 1 second.
But only in the rtp steam from me to the caller.
The second calls seems equal of timing.
Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no.
Now my question to you guys.
How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc).
Only capturing on one place don't did the trick for me.
Thanks for your hints
Cheers Karsten _______________________________________________ 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
Have you already taken a look at traces? Have you checked FS's MOS score, jitter, etc?
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Thu, Feb 6, 2020 at 6:43 AM Karsten Horsmann khorsmann@gmail.com wrote:
Hi yuriy,
Of course per default Kamailio don't handle rtp. I forgot that I use rtpengine on the "sbc" with Kamailio.
And as you all know, if it's sounds bad, then we sip guys get the tickets...
So that's an question for your experience in this area.
Cheers Karsten
Yuriy Gorlichenko ovoshlook@gmail.com schrieb am Do., 6. Feb. 2020, 10:27:
Kamailio does not handle RTP.
But you can use heplify ( or homer) That will alliw you to collect rtcp in your infrastructure and map it with the SIP calls you have.
On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, khorsmann@gmail.com wrote:
Hi List,
I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing.
So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios.
For testing purposes I setup an freeswitch with beep and then echo application to the caller.
My call flow are like this
Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together.
This generates for the caller an beep and echo.
The interesting thing is now, that Call 1 gets an hearable delay of 1 second.
But only in the rtp steam from me to the caller.
The second calls seems equal of timing.
Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no.
Now my question to you guys.
How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc).
Only capturing on one place don't did the trick for me.
Thanks for your hints
Cheers Karsten _______________________________________________ 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
Hi David,
I have a lot of traces. Today I bought in an cubro Tapping device to capture all rtp between internet and as second point the b2bua.
But Wireshark is not helpful on all calls, if the corresponding sip signaling is out of the captures (I have that one the inside).
It's a kind of black box testing to figure out where 386ms delay in one rtp direction can come. The isp showed me his MOS and jitter on there side and it was good. But it's an delay in the call 1 direction.
Tommorrow is another great delay hunting day.
Cheers Karsten
David Villasmil david.villasmil.work@gmail.com schrieb am Do., 6. Feb. 2020, 18:42:
Have you already taken a look at traces? Have you checked FS's MOS score, jitter, etc?
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Thu, Feb 6, 2020 at 6:43 AM Karsten Horsmann khorsmann@gmail.com wrote:
Hi yuriy,
Of course per default Kamailio don't handle rtp. I forgot that I use rtpengine on the "sbc" with Kamailio.
And as you all know, if it's sounds bad, then we sip guys get the tickets...
So that's an question for your experience in this area.
Cheers Karsten
Yuriy Gorlichenko ovoshlook@gmail.com schrieb am Do., 6. Feb. 2020, 10:27:
Kamailio does not handle RTP.
But you can use heplify ( or homer) That will alliw you to collect rtcp in your infrastructure and map it with the SIP calls you have.
On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, khorsmann@gmail.com wrote:
Hi List,
I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing.
So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios.
For testing purposes I setup an freeswitch with beep and then echo application to the caller.
My call flow are like this
Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together.
This generates for the caller an beep and echo.
The interesting thing is now, that Call 1 gets an hearable delay of 1 second.
But only in the rtp steam from me to the caller.
The second calls seems equal of timing.
Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no.
Now my question to you guys.
How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc).
Only capturing on one place don't did the trick for me.
Thanks for your hints
Cheers Karsten _______________________________________________ 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
-
David Villasmil -
Karsten Horsmann -
Yuriy Gorlichenko