Re: [Freeswitch-users] Session recording performance
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter. I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM. On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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Just curious, have you tried switching the boxes and seeing if you get same situation? Robert -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.... On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already. I'm still analyzing it, trying to find the reason for the jitter. On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert <rmundkowsky@ets.org> wrote:
Just curious, have you tried switching the boxes and seeing if you get same situation?
Robert
-----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance
but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call.
The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch....
On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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Any leads maybe under ethtool -S [interface] , possible some kind of checksum / network "interference" , possibly the tx side is doing more work? https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already.
I'm still analyzing it, trying to find the reason for the jitter.
On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert <rmundkowsky@ets.org> wrote:
Just curious, have you tried switching the boxes and seeing if you get same situation?
Robert
-----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance
but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call.
The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch....
On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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-- Sincerely, Brandon Armstead CTO / CRYY.com
but the other box is doing the same TX job, and it's doing fine. I'm trying different timers now. On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead <brandon@cryy.com> wrote:
Any leads maybe under ethtool -S [interface] , possible some kind of checksum / network "interference" , possibly the tx side is doing more work?
https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt
On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already.
I'm still analyzing it, trying to find the reason for the jitter.
On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert <rmundkowsky@ets.org> wrote:
Just curious, have you tried switching the boxes and seeing if you get same situation?
Robert
-----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance
but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call.
The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch....
On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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Your positive GSO / TSO / all are the same? Same Kernel? Same hardware? replica? What happens if you docker your test on 1 same physical host? On Thu, Sep 21, 2017 at 8:25 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
but the other box is doing the same TX job, and it's doing fine. I'm trying different timers now.
On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead <brandon@cryy.com> wrote:
Any leads maybe under ethtool -S [interface] , possible some kind of checksum / network "interference" , possibly the tx side is doing more work?
https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt
On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already.
I'm still analyzing it, trying to find the reason for the jitter.
On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert <rmundkowsky@ets.org> wrote:
Just curious, have you tried switching the boxes and seeing if you get same situation?
Robert
-----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance
but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call.
The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch....
On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote:
What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.
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it is two identical boxes, set up identically by myself :) I added mod_fdtimer, and set fdtimer as Sofia RTP timer, and the results are impressively better. I'll do some additional tests before a final commit. On Thu, Sep 21, 2017 at 5:28 PM, Brandon Armstead <brandon@cryy.com> wrote:
Your positive GSO / TSO / all are the same? Same Kernel? Same hardware? replica? What happens if you docker your test on 1 same physical host?
On Thu, Sep 21, 2017 at 8:25 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
but the other box is doing the same TX job, and it's doing fine. I'm trying different timers now.
On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead <brandon@cryy.com> wrote:
Any leads maybe under ethtool -S [interface] , possible some kind of checksum / network "interference" , possibly the tx side is doing more work?
https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt
On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already.
I'm still analyzing it, trying to find the reason for the jitter.
On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert <rmundkowsky@ets.org> wrote:
Just curious, have you tried switching the boxes and seeing if you get same situation?
Robert
-----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance
but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call.
The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch....
On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter.
I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM.
On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote: > What occurs if you send identical audio for the same duration? IOW, > play the sound file for each call. > > > _____________________________________________________________________ > ____ Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f > reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 > ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% > 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 > hWg%3D&reserved=0 > > Official FreeSWITCH Sites > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f > reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba > 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 > 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re > served=0 > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl > uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba > 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C > 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% > 3D&reserved=0 > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c > luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 > 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 > 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser > ved=0 > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists > .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% > 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 > 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT > vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > ers > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f > reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba > 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 > 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re > served=0
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswit...
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Thank you for your compliance.
________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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that's right, loading mod_timerfd, and setting in the SIP profile <param name="rtp-timer-name" value="timerfd"/> has eliminated the problem, and now all outbound RTP has jitter below 1ms. On Thu, Sep 21, 2017 at 6:06 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
it is two identical boxes, set up identically by myself :)
I added mod_fdtimer, and set fdtimer as Sofia RTP timer, and the results are impressively better. I'll do some additional tests before a final commit.
On Thu, Sep 21, 2017 at 5:28 PM, Brandon Armstead <brandon@cryy.com> wrote:
Your positive GSO / TSO / all are the same? Same Kernel? Same hardware? replica? What happens if you docker your test on 1 same physical host?
On Thu, Sep 21, 2017 at 8:25 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
but the other box is doing the same TX job, and it's doing fine. I'm trying different timers now.
On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead <brandon@cryy.com> wrote:
Any leads maybe under ethtool -S [interface] , possible some kind of checksum / network "interference" , possibly the tx side is doing more work?
https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt
On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already.
I'm still analyzing it, trying to find the reason for the jitter.
On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert <rmundkowsky@ets.org> wrote:
Just curious, have you tried switching the boxes and seeing if you get same situation?
Robert
-----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Session recording performance
but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call.
The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch....
On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote: > with jitterbuffer, it now works as expected. So, the sending party was > generating extra jitter. > > I am playing the same audio file in both directions, and then compare > the recorded audio with the original by Sevana AQuA software. This > allows me to detect packet loss in a heterogeneous environment where > the voice goes through RTP proxies and TDM. > > > > > On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0@gmail.com> wrote: >> What occurs if you send identical audio for the same duration? IOW, >> play the sound file for each call. >> >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >> hWg%3D&reserved=0 >> >> Official FreeSWITCH Sites >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >> served=0 >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >> 3D&reserved=0 >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >> ved=0 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >> served=0
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Isn't timerfd the default under-the-hood implementation of the "soft" (the default) timer in FS for some years now? In other words, it's quite surprising this param had any effect at all. Are you sure it did it? What are your kernel, glibc (ldd --version) and FS versions? *From:* Stanislav Sinyagin *Sent:* Thursday, September 21, 2017 16:55 *To:* Freeswitch Users Help *Subject:* Re: [Freeswitch-users] Session recording performance that's right, loading mod_timerfd, and setting in the SIP profile <param name="rtp-timer-name" value="timerfd"/> has eliminated the problem, and now all outbound RTP has jitter below 1ms.
I thought so too, but I detected a high jitter in outbound RTP (over 10ms) , and this parameter has resolved it. The installation is as follows: The physical host is running under Debian Stretch, kernel version 4.9.0-3-amd64. The VoIP probe is isolated into an LXC container, and the container is running Debian Jessie. The container is not limited in any resources, and FreeSWITCH receives its real-time priority as it should. I tested the priority explicitly by running other processes in normal priority, and their load has no effect on FreeSWITCH performance. On the physical host: ldd (Debian GLIBC 2.24-11+deb9u1) 2.24 Inside the container: ldd (Debian GLIBC 2.19-18+deb8u10) 2.19 freeswitch@fsnic2> status UP 0 years, 0 days, 5 hours, 25 minutes, 25 seconds, 101 milliseconds, 11 microseconds FreeSWITCH (Version 1.6.19 -36-7a77e0b 64bit) is ready 12924 session(s) since startup 16 session(s) - peak 17, last 5min 17 1 session(s) per Sec out of max 30, peak 1, last 5min 1 1000 session(s) max min idle cpu 0.00/89.70 Current Stack Size/Max 240K/8192K So, the only difference from a standard baremetal installation, is a newer kernel version. But all other libraries are from Jessie. I will eventually make additional tests with baremetal servers without LXC, but I don't think it's the source of the issue. On Fri, Sep 22, 2017 at 2:30 AM, Anatoli <me@anatoli.ws> wrote:
Isn't timerfd the default under-the-hood implementation of the "soft" (the default) timer in FS for some years now? In other words, it's quite surprising this param had any effect at all. Are you sure it did it? What are your kernel, glibc (ldd --version) and FS versions?
From: Stanislav Sinyagin Sent: Thursday, September 21, 2017 16:55 To: Freeswitch Users Help Subject: Re: [Freeswitch-users] Session recording performance
that's right, loading mod_timerfd, and setting in the SIP profile <param name="rtp-timer-name" value="timerfd"/>
has eliminated the problem, and now all outbound RTP has jitter below 1ms.
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participants (5)
-
Anatoli
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Brandon Armstead
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jungle Boogie
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Mundkowsky, Robert
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Stanislav Sinyagin