Hi, I'm using latest git-version of Freeswitch, and when I go to voicemail when calling a number the sound playback is choppy and it skips some of the digits in the number I called. If I set sleep to 2000 it sounds better, but it's still a little choppy and still skips some of the digits. This is with the default settings coming with the git version. When I call 4000 to listen to voicemail it sounds fine. The kernel is the default kernel which is compiled with CONFIG_HZ_250=y Regards Oyvind
Do you realize the latest GIT is an hourly thing? What revision do you mean by latest? We do not support anything but 1000hz kernels our platform of choice is CentOS 5.2/5.3 Anything else you are experimenting...... On Fri, Jan 28, 2011 at 6:39 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail when calling a number the sound playback is choppy and it skips some of the digits in the number I called.
If I set sleep to 2000 it sounds better, but it's still a little choppy and still skips some of the digits.
This is with the default settings coming with the git version. When I call 4000 to listen to voicemail it sounds fine.
The kernel is the default kernel which is compiled with CONFIG_HZ_250=y
Regards Oyvind
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail when calling a number the sound playback is choppy and it skips some of the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works. HTH --FC
Frank, I fail to see the relationship between the hw timer and NTP. Can you please elaborate ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail when calling a number the sound playback is choppy and it skips some of the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone <david.ponzone@ipeva.fr>wrote:
Frank,
I fail to see the relationship between the hw timer and NTP. Can you please elaborate ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
*Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. **IPeva** décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.* * *
Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail
when calling a number the sound playback is choppy and it skips some of
the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
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-- ----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue. On Sat, Jan 29, 2011 at 2:12 PM, Frank Park <frank@telonium.com> wrote:
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank
On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
Frank, I fail to see the relationship between the hw timer and NTP. Can you please elaborate ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail
when calling a number the sound playback is choppy and it skips some of
the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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--
----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thanks for all your feedback. I'll keep on trying and inform you what worked for me. Regards Oyvind On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank
On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> wrote:
Frank, I fail to see the relationship between the hw timer and NTP. Can you please elaborate ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail
when calling a number the sound playback is choppy and it skips some of
the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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--
----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel). We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process. For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though. On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank
On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> wrote:
Frank, I fail to see the relationship between the hw timer and NTP. Can you please elaborate ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail
when calling a number the sound playback is choppy and it skips some of
the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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--
----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
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Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side. If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it. On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank
On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> wrote:
Frank, I fail to see the relationship between the hw timer and NTP. Can you please elaborate ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail
when calling a number the sound playback is choppy and it skips some of
the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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--
----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Yes under windows the most recent commit causes a severe deterioration in the audio. I had thought it may only be a problem with windows but I didnt really know. I have added some debug to http://jira.freeswitch.org/browse/FS-3011 with some timing information under windows that illustrates the problem. I can provide more information if needed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choppy-VM-when-using-loopback-... Sent from the freeswitch-users mailing list archive at Nabble.com.
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes. I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue. I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there. On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank
On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> wrote:
Frank, I fail to see the relationship between the hw timer and NTP. Can you please elaborate ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/01/2011 à 19:49, Frank Carmickle a écrit :
Hi
On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote:
Hi,
I'm using latest git-version of Freeswitch, and when I go to voicemail
when calling a number the sound playback is choppy and it skips some of
the digits in the number I called.
What kind of results do you get from timer_test at the fs_cli? Are you running on hardware or are you virtualized? What is your clock source set to and what are your available clock source options? See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource. I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me. When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between. I have my clocksource set to jiffies and xen independent wallclock set to 1. Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady. I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying. If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems. Just play with the different timer options until you find the one that works.
HTH --FC
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--
----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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Try the latest GIT, I reverted the last patch and tried to solve the problem differently. On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
Yeah. I, too, don't see the correlation between the NTP and hw timer.. I am not familiar with the timer_test command and what it's measuring, but of the 50 tests it ran, min is 19089 and max is 20713. Frank
On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> wrote:
> Frank, > I fail to see the relationship between the hw timer and NTP. > Can you please elaborate ? > David Ponzone Direction Technique > email: david.ponzone@ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > Ce message et toutes les pièces jointes sont confidentiels et établis à > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autorisée est interdite. Tout message électronique est susceptible > d'altération. IPeva décline toute responsabilité au titre de ce message s'il > a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce > message, merci de le détruire immédiatement et d'avertir l'expéditeur. > > > > Le 28/01/2011 à 19:49, Frank Carmickle a écrit : > > Hi > > On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: > > Hi, > > I'm using latest git-version of Freeswitch, and when I go to voicemail > > when calling a number the sound playback is choppy and it skips some of > > the digits in the number I called. > > What kind of results do you get from timer_test at the fs_cli? Are you > running on hardware or are you virtualized? What is your clock source set > to and what are your available clock source options? See > /sys/devices/system/clocksource/clocksource0/available_clocksource and > /sys/devices/system/clocksource/clocksource0/current_clocksource. I am > running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to > hang at 19998/19999 which works very well for me. When I was having problem > it was reporting numbers all over the map from 17400 to 22600 with lots of > randomness in between. I have my clocksource set to jiffies and xen > independent wallclock set to 1. Of course at that point you need to have > ntp running against a bunch of servers to drive your clock nice and steady. > I know my set up is probably a lot different than yours but I thought I'd > toss it out there to show that some of the harshest conditions can be dealt > with and don't give up trying. If you are running on hardware with a cpu > that doesn't have constant_tsc then you might have some problems. Just play > with the different timer options until you find the one that works. > > HTH > --FC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
--
----=======================---- Frank Park Telonium Communications, LLC frank@telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I've now tested the latest GIT, and it now works fine with the Debian lenny default kernel. I didnt get around to recompiling it for 1000 HZ though, but I tried changing clocksource which didnt help at all. This is on a Dell server, and I'm not running it on a virtual server. FYI I also tried latest GIT yesterday, with the same results as stated in my first mail. Regards Oyvind On 2011-02-01 21:47, Anthony Minessale wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane<null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane<null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen<oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
> Yeah. I, too, don't see the correlation between the NTP and hw timer.. > I am not familiar with the timer_test command and what it's measuring, but > of the 50 tests it ran, min is 19089 and max is 20713. > Frank > > > On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> > wrote: > > >> Frank, >> I fail to see the relationship between the hw timer and NTP. >> Can you please elaborate ? >> David Ponzone Direction Technique >> email: david.ponzone@ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> Ce message et toutes les pièces jointes sont confidentiels et établis à >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autorisée est interdite. Tout message électronique est susceptible >> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >> >> >> >> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >> >> Hi >> >> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >> >> Hi, >> >> I'm using latest git-version of Freeswitch, and when I go to voicemail >> >> when calling a number the sound playback is choppy and it skips some of >> >> the digits in the number I called. >> >> What kind of results do you get from timer_test at the fs_cli? Are you >> running on hardware or are you virtualized? What is your clock source set >> to and what are your available clock source options? See >> /sys/devices/system/clocksource/clocksource0/available_clocksource and >> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >> hang at 19998/19999 which works very well for me. When I was having problem >> it was reporting numbers all over the map from 17400 to 22600 with lots of >> randomness in between. I have my clocksource set to jiffies and xen >> independent wallclock set to 1. Of course at that point you need to have >> ntp running against a bunch of servers to drive your clock nice and steady. >> I know my set up is probably a lot different than yours but I thought I'd >> toss it out there to show that some of the harshest conditions can be dealt >> with and don't give up trying. If you are running on hardware with a cpu >> that doesn't have constant_tsc then you might have some problems. Just play >> with the different timer options until you find the one that works. >> >> HTH >> --FC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > -- > > ----=======================---- > Frank Park > Telonium Communications, LLC > frank@telonium.com > http://www.telonium.com > Follow Us on Twitter: @GetTelonium > 404-566-8888 x1001 Office > 404-939-4242 Cell > ----=======================---- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > >
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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This seems to have resolved the issue for us :) Thanks. On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote:
Everyone should try latest GIT before pondering any further because I added a patch like 2 days ago to adress this issue.
On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote:
> Yeah. I, too, don't see the correlation between the NTP and hw timer.. > I am not familiar with the timer_test command and what it's measuring, but > of the 50 tests it ran, min is 19089 and max is 20713. > Frank > > > On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> > wrote: > >> Frank, >> I fail to see the relationship between the hw timer and NTP. >> Can you please elaborate ? >> David Ponzone Direction Technique >> email: david.ponzone@ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> Ce message et toutes les pièces jointes sont confidentiels et établis à >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autorisée est interdite. Tout message électronique est susceptible >> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >> >> >> >> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >> >> Hi >> >> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >> >> Hi, >> >> I'm using latest git-version of Freeswitch, and when I go to voicemail >> >> when calling a number the sound playback is choppy and it skips some of >> >> the digits in the number I called. >> >> What kind of results do you get from timer_test at the fs_cli? Are you >> running on hardware or are you virtualized? What is your clock source set >> to and what are your available clock source options? See >> /sys/devices/system/clocksource/clocksource0/available_clocksource and >> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >> hang at 19998/19999 which works very well for me. When I was having problem >> it was reporting numbers all over the map from 17400 to 22600 with lots of >> randomness in between. I have my clocksource set to jiffies and xen >> independent wallclock set to 1. Of course at that point you need to have >> ntp running against a bunch of servers to drive your clock nice and steady. >> I know my set up is probably a lot different than yours but I thought I'd >> toss it out there to show that some of the harshest conditions can be dealt >> with and don't give up trying. If you are running on hardware with a cpu >> that doesn't have constant_tsc then you might have some problems. Just play >> with the different timer options until you find the one that works. >> >> HTH >> --FC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ----=======================---- > Frank Park > Telonium Communications, LLC > frank@telonium.com > http://www.telonium.com > Follow Us on Twitter: @GetTelonium > 404-566-8888 x1001 Office > 404-939-4242 Cell > ----=======================---- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > >
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-- Anthony Minessale II
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-- Anthony Minessale II
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AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy. We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false 2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote:
Thanks for all your feedback. I'll keep on trying and inform you what worked for me.
Regards Oyvind
On 2011-01-29 21:48, Anthony Minessale wrote: > Everyone should try latest GIT before pondering any further because I > added a patch like 2 days ago to adress this issue. > > > On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: > >> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >> I am not familiar with the timer_test command and what it's measuring, but >> of the 50 tests it ran, min is 19089 and max is 20713. >> Frank >> >> >> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >> wrote: >> >>> Frank, >>> I fail to see the relationship between the hw timer and NTP. >>> Can you please elaborate ? >>> David Ponzone Direction Technique >>> email: david.ponzone@ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autorisée est interdite. Tout message électronique est susceptible >>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>> >>> >>> >>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>> >>> Hi >>> >>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>> >>> Hi, >>> >>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>> >>> when calling a number the sound playback is choppy and it skips some of >>> >>> the digits in the number I called. >>> >>> What kind of results do you get from timer_test at the fs_cli? Are you >>> running on hardware or are you virtualized? What is your clock source set >>> to and what are your available clock source options? See >>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>> hang at 19998/19999 which works very well for me. When I was having problem >>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>> randomness in between. I have my clocksource set to jiffies and xen >>> independent wallclock set to 1. Of course at that point you need to have >>> ntp running against a bunch of servers to drive your clock nice and steady. >>> I know my set up is probably a lot different than yours but I thought I'd >>> toss it out there to show that some of the harshest conditions can be dealt >>> with and don't give up trying. If you are running on hardware with a cpu >>> that doesn't have constant_tsc then you might have some problems. Just play >>> with the different timer options until you find the one that works. >>> >>> HTH >>> --FC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> >> ----=======================---- >> Frank Park >> Telonium Communications, LLC >> frank@telonium.com >> http://www.telonium.com >> Follow Us on Twitter: @GetTelonium >> 404-566-8888 x1001 Office >> 404-939-4242 Cell >> ----=======================---- >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > >
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Best regards, Dmitry Sytchev, IT Engineer
commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b Author: Anthony Minessale <anthm@freeswitch.org> Date: Tue Feb 1 16:22:36 2011 -0600 this is the one you need at a bare minimum. If you are still having problems the most likely cause is ptime mismatch between the 2 ends of the call. you cannot leave loopback in the call path when the legs are not on the same ptime with very good results. On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy.
We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false
2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote:
FWIW we've been also been having audio issues with loopback recently on EC2 (with a 1000Hz kernel).
We worked around it in the short term by reverting mod_loopback to git-4c5426f during the build process.
For anyone else who wants to try this just run "git checkout 4c5426f" in src/mod/endpoints/mod_loopback then build as usual. This is NOT a long term solution though.
On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote: > Thanks for all your feedback. I'll keep on trying and inform you what > worked for me. > > > Regards > Oyvind > > On 2011-01-29 21:48, Anthony Minessale wrote: >> Everyone should try latest GIT before pondering any further because I >> added a patch like 2 days ago to adress this issue. >> >> >> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: >> >>> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >>> I am not familiar with the timer_test command and what it's measuring, but >>> of the 50 tests it ran, min is 19089 and max is 20713. >>> Frank >>> >>> >>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >>> wrote: >>> >>>> Frank, >>>> I fail to see the relationship between the hw timer and NTP. >>>> Can you please elaborate ? >>>> David Ponzone Direction Technique >>>> email: david.ponzone@ipeva.fr >>>> tel: 01 74 03 18 97 >>>> gsm: 06 66 98 76 34 >>>> Service Client IPeva >>>> tel: 0811 46 26 26 >>>> www.ipeva.fr - www.ipeva-studio.com >>>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>> non autorisée est interdite. Tout message électronique est susceptible >>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>>> >>>> >>>> >>>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>>> >>>> Hi >>>> >>>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>>> >>>> Hi, >>>> >>>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>>> >>>> when calling a number the sound playback is choppy and it skips some of >>>> >>>> the digits in the number I called. >>>> >>>> What kind of results do you get from timer_test at the fs_cli? Are you >>>> running on hardware or are you virtualized? What is your clock source set >>>> to and what are your available clock source options? See >>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>>> hang at 19998/19999 which works very well for me. When I was having problem >>>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>>> randomness in between. I have my clocksource set to jiffies and xen >>>> independent wallclock set to 1. Of course at that point you need to have >>>> ntp running against a bunch of servers to drive your clock nice and steady. >>>> I know my set up is probably a lot different than yours but I thought I'd >>>> toss it out there to show that some of the harshest conditions can be dealt >>>> with and don't give up trying. If you are running on hardware with a cpu >>>> that doesn't have constant_tsc then you might have some problems. Just play >>>> with the different timer options until you find the one that works. >>>> >>>> HTH >>>> --FC >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> >>> ----=======================---- >>> Frank Park >>> Telonium Communications, LLC >>> frank@telonium.com >>> http://www.telonium.com >>> Follow Us on Twitter: @GetTelonium >>> 404-566-8888 x1001 Office >>> 404-939-4242 Cell >>> ----=======================---- >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thanks, I will test tomorrow. I'm wondering if I could use att_xfer with loopbacks after these patches:D 2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b Author: Anthony Minessale <anthm@freeswitch.org> Date: Tue Feb 1 16:22:36 2011 -0600
this is the one you need at a bare minimum.
If you are still having problems the most likely cause is ptime mismatch between the 2 ends of the call. you cannot leave loopback in the call path when the legs are not on the same ptime with very good results.
On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy.
We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false
2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you saying you have better results on that version than you do on the latest? What conditions do you have that cause you trouble, what is the endpoint on the other side.
If the last commit to mod_loopback intended to improve audio quality actually makes it worse I need to investigate it.
On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote: > FWIW we've been also been having audio issues with loopback recently > on EC2 (with a 1000Hz kernel). > > We worked around it in the short term by reverting mod_loopback to > git-4c5426f during the build process. > > For anyone else who wants to try this just run "git checkout 4c5426f" > in src/mod/endpoints/mod_loopback then build as usual. This is NOT a > long term solution though. > > On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote: >> Thanks for all your feedback. I'll keep on trying and inform you what >> worked for me. >> >> >> Regards >> Oyvind >> >> On 2011-01-29 21:48, Anthony Minessale wrote: >>> Everyone should try latest GIT before pondering any further because I >>> added a patch like 2 days ago to adress this issue. >>> >>> >>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: >>> >>>> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >>>> I am not familiar with the timer_test command and what it's measuring, but >>>> of the 50 tests it ran, min is 19089 and max is 20713. >>>> Frank >>>> >>>> >>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >>>> wrote: >>>> >>>>> Frank, >>>>> I fail to see the relationship between the hw timer and NTP. >>>>> Can you please elaborate ? >>>>> David Ponzone Direction Technique >>>>> email: david.ponzone@ipeva.fr >>>>> tel: 01 74 03 18 97 >>>>> gsm: 06 66 98 76 34 >>>>> Service Client IPeva >>>>> tel: 0811 46 26 26 >>>>> www.ipeva.fr - www.ipeva-studio.com >>>>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>> non autorisée est interdite. Tout message électronique est susceptible >>>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>>>> >>>>> >>>>> >>>>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>>>> >>>>> Hi >>>>> >>>>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>>>> >>>>> Hi, >>>>> >>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>>>> >>>>> when calling a number the sound playback is choppy and it skips some of >>>>> >>>>> the digits in the number I called. >>>>> >>>>> What kind of results do you get from timer_test at the fs_cli? Are you >>>>> running on hardware or are you virtualized? What is your clock source set >>>>> to and what are your available clock source options? See >>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>>>> hang at 19998/19999 which works very well for me. When I was having problem >>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>>>> randomness in between. I have my clocksource set to jiffies and xen >>>>> independent wallclock set to 1. Of course at that point you need to have >>>>> ntp running against a bunch of servers to drive your clock nice and steady. >>>>> I know my set up is probably a lot different than yours but I thought I'd >>>>> toss it out there to show that some of the harshest conditions can be dealt >>>>> with and don't give up trying. If you are running on hardware with a cpu >>>>> that doesn't have constant_tsc then you might have some problems. Just play >>>>> with the different timer options until you find the one that works. >>>>> >>>>> HTH >>>>> --FC >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> >>>> ----=======================---- >>>> Frank Park >>>> Telonium Communications, LLC >>>> frank@telonium.com >>>> http://www.telonium.com >>>> Follow Us on Twitter: @GetTelonium >>>> 404-566-8888 x1001 Office >>>> 404-939-4242 Cell >>>> ----=======================---- >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> >>> >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Best regards,
Dmitry Sytchev, IT Engineer
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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-- Best regards, Dmitry Sytchev, IT Engineer
maybe, but using loopback without the bowout is very costly. you can also consider hair-pinning a call back to your own box over SIP for this type of thing. On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
Thanks, I will test tomorrow.
I'm wondering if I could use att_xfer with loopbacks after these patches:D
2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b Author: Anthony Minessale <anthm@freeswitch.org> Date: Tue Feb 1 16:22:36 2011 -0600
this is the one you need at a bare minimum.
If you are still having problems the most likely cause is ptime mismatch between the 2 ends of the call. you cannot leave loopback in the call path when the legs are not on the same ptime with very good results.
On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy.
We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false
2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote:
Yes, using 1f1541b our calls that use mod_loopback into mod_conference are unusable. The audio is choppy and delay increases as time passes.
I thought it was an issue relating to the default 100Hz kernel on EC2 so I spent some time yesterday putting together a 1000Hz kernel but it didn't make any difference. In the meantime I've compiled mod_loopback from 4c5426f and loaded it with my 1f1541b build which eliminates the issue.
I haven't added it to Jira yet as I want to spend some time debugging it (and I also owe you some debug info for FS-2934) but the problem is definitely there.
On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > Are you saying you have better results on that version than you do on > the latest? > What conditions do you have that cause you trouble, what is the > endpoint on the other side. > > If the last commit to mod_loopback intended to improve audio quality > actually makes it worse I need to investigate it. > > > On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote: >> FWIW we've been also been having audio issues with loopback recently >> on EC2 (with a 1000Hz kernel). >> >> We worked around it in the short term by reverting mod_loopback to >> git-4c5426f during the build process. >> >> For anyone else who wants to try this just run "git checkout 4c5426f" >> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a >> long term solution though. >> >> On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote: >>> Thanks for all your feedback. I'll keep on trying and inform you what >>> worked for me. >>> >>> >>> Regards >>> Oyvind >>> >>> On 2011-01-29 21:48, Anthony Minessale wrote: >>>> Everyone should try latest GIT before pondering any further because I >>>> added a patch like 2 days ago to adress this issue. >>>> >>>> >>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: >>>> >>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >>>>> I am not familiar with the timer_test command and what it's measuring, but >>>>> of the 50 tests it ran, min is 19089 and max is 20713. >>>>> Frank >>>>> >>>>> >>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >>>>> wrote: >>>>> >>>>>> Frank, >>>>>> I fail to see the relationship between the hw timer and NTP. >>>>>> Can you please elaborate ? >>>>>> David Ponzone Direction Technique >>>>>> email: david.ponzone@ipeva.fr >>>>>> tel: 01 74 03 18 97 >>>>>> gsm: 06 66 98 76 34 >>>>>> Service Client IPeva >>>>>> tel: 0811 46 26 26 >>>>>> www.ipeva.fr - www.ipeva-studio.com >>>>>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>>> non autorisée est interdite. Tout message électronique est susceptible >>>>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>>>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>>>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>>>>> >>>>>> >>>>>> >>>>>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>>>>> >>>>>> Hi >>>>>> >>>>>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>>>>> >>>>>> when calling a number the sound playback is choppy and it skips some of >>>>>> >>>>>> the digits in the number I called. >>>>>> >>>>>> What kind of results do you get from timer_test at the fs_cli? Are you >>>>>> running on hardware or are you virtualized? What is your clock source set >>>>>> to and what are your available clock source options? See >>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>>>>> hang at 19998/19999 which works very well for me. When I was having problem >>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>>>>> randomness in between. I have my clocksource set to jiffies and xen >>>>>> independent wallclock set to 1. Of course at that point you need to have >>>>>> ntp running against a bunch of servers to drive your clock nice and steady. >>>>>> I know my set up is probably a lot different than yours but I thought I'd >>>>>> toss it out there to show that some of the harshest conditions can be dealt >>>>>> with and don't give up trying. If you are running on hardware with a cpu >>>>>> that doesn't have constant_tsc then you might have some problems. Just play >>>>>> with the different timer options until you find the one that works. >>>>>> >>>>>> HTH >>>>>> --FC >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> ----=======================---- >>>>> Frank Park >>>>> Telonium Communications, LLC >>>>> frank@telonium.com >>>>> http://www.telonium.com >>>>> Follow Us on Twitter: @GetTelonium >>>>> 404-566-8888 x1001 Office >>>>> 404-939-4242 Cell >>>>> ----=======================---- >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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-- Best regards,
Dmitry Sytchev, IT Engineer
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
also consider passing {rtp_timer_name=none} through to disable the rtp timer in the sip leg when butting up against mod_loopback On Thu, Feb 3, 2011 at 12:32 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
maybe, but using loopback without the bowout is very costly. you can also consider hair-pinning a call back to your own box over SIP for this type of thing.
On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
Thanks, I will test tomorrow.
I'm wondering if I could use att_xfer with loopbacks after these patches:D
2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b Author: Anthony Minessale <anthm@freeswitch.org> Date: Tue Feb 1 16:22:36 2011 -0600
this is the one you need at a bare minimum.
If you are still having problems the most likely cause is ptime mismatch between the 2 ends of the call. you cannot leave loopback in the call path when the legs are not on the same ptime with very good results.
On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy.
We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false
2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote: > Yes, using 1f1541b our calls that use mod_loopback into mod_conference > are unusable. The audio is choppy and delay increases as time passes. > > I thought it was an issue relating to the default 100Hz kernel on EC2 > so I spent some time yesterday putting together a 1000Hz kernel but it > didn't make any difference. In the meantime I've compiled mod_loopback > from 4c5426f and loaded it with my 1f1541b build which eliminates the > issue. > > I haven't added it to Jira yet as I want to spend some time debugging > it (and I also owe you some debug info for FS-2934) but the problem is > definitely there. > > On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale > <anthony.minessale@gmail.com> wrote: >> Are you saying you have better results on that version than you do on >> the latest? >> What conditions do you have that cause you trouble, what is the >> endpoint on the other side. >> >> If the last commit to mod_loopback intended to improve audio quality >> actually makes it worse I need to investigate it. >> >> >> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote: >>> FWIW we've been also been having audio issues with loopback recently >>> on EC2 (with a 1000Hz kernel). >>> >>> We worked around it in the short term by reverting mod_loopback to >>> git-4c5426f during the build process. >>> >>> For anyone else who wants to try this just run "git checkout 4c5426f" >>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a >>> long term solution though. >>> >>> On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote: >>>> Thanks for all your feedback. I'll keep on trying and inform you what >>>> worked for me. >>>> >>>> >>>> Regards >>>> Oyvind >>>> >>>> On 2011-01-29 21:48, Anthony Minessale wrote: >>>>> Everyone should try latest GIT before pondering any further because I >>>>> added a patch like 2 days ago to adress this issue. >>>>> >>>>> >>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: >>>>> >>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >>>>>> I am not familiar with the timer_test command and what it's measuring, but >>>>>> of the 50 tests it ran, min is 19089 and max is 20713. >>>>>> Frank >>>>>> >>>>>> >>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >>>>>> wrote: >>>>>> >>>>>>> Frank, >>>>>>> I fail to see the relationship between the hw timer and NTP. >>>>>>> Can you please elaborate ? >>>>>>> David Ponzone Direction Technique >>>>>>> email: david.ponzone@ipeva.fr >>>>>>> tel: 01 74 03 18 97 >>>>>>> gsm: 06 66 98 76 34 >>>>>>> Service Client IPeva >>>>>>> tel: 0811 46 26 26 >>>>>>> www.ipeva.fr - www.ipeva-studio.com >>>>>>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>>>> non autorisée est interdite. Tout message électronique est susceptible >>>>>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>>>>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>>>>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>>>>>> >>>>>>> >>>>>>> >>>>>>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>>>>>> >>>>>>> Hi >>>>>>> >>>>>>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>>>>>> >>>>>>> when calling a number the sound playback is choppy and it skips some of >>>>>>> >>>>>>> the digits in the number I called. >>>>>>> >>>>>>> What kind of results do you get from timer_test at the fs_cli? Are you >>>>>>> running on hardware or are you virtualized? What is your clock source set >>>>>>> to and what are your available clock source options? See >>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>>>>>> hang at 19998/19999 which works very well for me. When I was having problem >>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>>>>>> randomness in between. I have my clocksource set to jiffies and xen >>>>>>> independent wallclock set to 1. Of course at that point you need to have >>>>>>> ntp running against a bunch of servers to drive your clock nice and steady. >>>>>>> I know my set up is probably a lot different than yours but I thought I'd >>>>>>> toss it out there to show that some of the harshest conditions can be dealt >>>>>>> with and don't give up trying. If you are running on hardware with a cpu >>>>>>> that doesn't have constant_tsc then you might have some problems. Just play >>>>>>> with the different timer options until you find the one that works. >>>>>>> >>>>>>> HTH >>>>>>> --FC >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> ----=======================---- >>>>>> Frank Park >>>>>> Telonium Communications, LLC >>>>>> frank@telonium.com >>>>>> http://www.telonium.com >>>>>> Follow Us on Twitter: @GetTelonium >>>>>> 404-566-8888 x1001 Office >>>>>> 404-939-4242 Cell >>>>>> ----=======================---- >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> googletalk:conf+888@conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Yes, I also had this idea, but I thought it would lead to troubles with loop detection. How this can be done exactly? I need separate profiles for this? Or I just can dial from internal to internal? And what if I use Kamailio in front of FS? Then need to deal with call re-routing to kamailio. I thought it may lead to spiral detection, if I understand right what spiral is. 2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
maybe, but using loopback without the bowout is very costly. you can also consider hair-pinning a call back to your own box over SIP for this type of thing.
On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
Thanks, I will test tomorrow.
I'm wondering if I could use att_xfer with loopbacks after these patches:D
2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b Author: Anthony Minessale <anthm@freeswitch.org> Date: Tue Feb 1 16:22:36 2011 -0600
this is the one you need at a bare minimum.
If you are still having problems the most likely cause is ptime mismatch between the 2 ends of the call. you cannot leave loopback in the call path when the legs are not on the same ptime with very good results.
On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy.
We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false
2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try the latest GIT, I reverted the last patch and tried to solve the problem differently.
On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote: > Yes, using 1f1541b our calls that use mod_loopback into mod_conference > are unusable. The audio is choppy and delay increases as time passes. > > I thought it was an issue relating to the default 100Hz kernel on EC2 > so I spent some time yesterday putting together a 1000Hz kernel but it > didn't make any difference. In the meantime I've compiled mod_loopback > from 4c5426f and loaded it with my 1f1541b build which eliminates the > issue. > > I haven't added it to Jira yet as I want to spend some time debugging > it (and I also owe you some debug info for FS-2934) but the problem is > definitely there. > > On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale > <anthony.minessale@gmail.com> wrote: >> Are you saying you have better results on that version than you do on >> the latest? >> What conditions do you have that cause you trouble, what is the >> endpoint on the other side. >> >> If the last commit to mod_loopback intended to improve audio quality >> actually makes it worse I need to investigate it. >> >> >> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote: >>> FWIW we've been also been having audio issues with loopback recently >>> on EC2 (with a 1000Hz kernel). >>> >>> We worked around it in the short term by reverting mod_loopback to >>> git-4c5426f during the build process. >>> >>> For anyone else who wants to try this just run "git checkout 4c5426f" >>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a >>> long term solution though. >>> >>> On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote: >>>> Thanks for all your feedback. I'll keep on trying and inform you what >>>> worked for me. >>>> >>>> >>>> Regards >>>> Oyvind >>>> >>>> On 2011-01-29 21:48, Anthony Minessale wrote: >>>>> Everyone should try latest GIT before pondering any further because I >>>>> added a patch like 2 days ago to adress this issue. >>>>> >>>>> >>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: >>>>> >>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >>>>>> I am not familiar with the timer_test command and what it's measuring, but >>>>>> of the 50 tests it ran, min is 19089 and max is 20713. >>>>>> Frank >>>>>> >>>>>> >>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >>>>>> wrote: >>>>>> >>>>>>> Frank, >>>>>>> I fail to see the relationship between the hw timer and NTP. >>>>>>> Can you please elaborate ? >>>>>>> David Ponzone Direction Technique >>>>>>> email: david.ponzone@ipeva.fr >>>>>>> tel: 01 74 03 18 97 >>>>>>> gsm: 06 66 98 76 34 >>>>>>> Service Client IPeva >>>>>>> tel: 0811 46 26 26 >>>>>>> www.ipeva.fr - www.ipeva-studio.com >>>>>>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>>>> non autorisée est interdite. Tout message électronique est susceptible >>>>>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>>>>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>>>>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>>>>>> >>>>>>> >>>>>>> >>>>>>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>>>>>> >>>>>>> Hi >>>>>>> >>>>>>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>>>>>> >>>>>>> when calling a number the sound playback is choppy and it skips some of >>>>>>> >>>>>>> the digits in the number I called. >>>>>>> >>>>>>> What kind of results do you get from timer_test at the fs_cli? Are you >>>>>>> running on hardware or are you virtualized? What is your clock source set >>>>>>> to and what are your available clock source options? See >>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>>>>>> hang at 19998/19999 which works very well for me. When I was having problem >>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>>>>>> randomness in between. I have my clocksource set to jiffies and xen >>>>>>> independent wallclock set to 1. Of course at that point you need to have >>>>>>> ntp running against a bunch of servers to drive your clock nice and steady. >>>>>>> I know my set up is probably a lot different than yours but I thought I'd >>>>>>> toss it out there to show that some of the harshest conditions can be dealt >>>>>>> with and don't give up trying. If you are running on hardware with a cpu >>>>>>> that doesn't have constant_tsc then you might have some problems. Just play >>>>>>> with the different timer options until you find the one that works. >>>>>>> >>>>>>> HTH >>>>>>> --FC >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> ----=======================---- >>>>>> Frank Park >>>>>> Telonium Communications, LLC >>>>>> frank@telonium.com >>>>>> http://www.telonium.com >>>>>> Follow Us on Twitter: @GetTelonium >>>>>> 404-566-8888 x1001 Office >>>>>> 404-939-4242 Cell >>>>>> ----=======================---- >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> googletalk:conf+888@conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards, Dmitry Sytchev, IT Engineer
you can just call sofia/profilename/ext@<local ip> even when its the same profile. On Thu, Feb 3, 2011 at 1:06 PM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
Yes, I also had this idea, but I thought it would lead to troubles with loop detection. How this can be done exactly? I need separate profiles for this? Or I just can dial from internal to internal?
And what if I use Kamailio in front of FS? Then need to deal with call re-routing to kamailio. I thought it may lead to spiral detection, if I understand right what spiral is.
2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
maybe, but using loopback without the bowout is very costly. you can also consider hair-pinning a call back to your own box over SIP for this type of thing.
On Thu, Feb 3, 2011 at 12:28 PM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
Thanks, I will test tomorrow.
I'm wondering if I could use att_xfer with loopbacks after these patches:D
2011/2/3 Anthony Minessale <anthony.minessale@gmail.com>:
commit fb66abfab4a74055c38cdc67da83e6e0175a4a0b Author: Anthony Minessale <anthm@freeswitch.org> Date: Tue Feb 1 16:22:36 2011 -0600
this is the one you need at a bare minimum.
If you are still having problems the most likely cause is ptime mismatch between the 2 ends of the call. you cannot leave loopback in the call path when the legs are not on the same ptime with very good results.
On Thu, Feb 3, 2011 at 4:31 AM, Dmitry Sytchev <kbdfck@gmail.com> wrote:
We faced same trouble with FreeSWITCH Version 1.0.head (git-33848eb 2011-02-01 08-56-26 +0100), the problem persists with loopback channels. Even if two sip endpoints are bridged via loopback, sound is choppy.
We have bridge_early_media=true on loopback channel bridge command and loopback_bowout=false, loopback_bowout_on_execute=false
2011/2/2 Dan Lane <null@invalid.name>:
This seems to have resolved the issue for us :)
Thanks.
On Tue, Feb 1, 2011 at 8:47 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > Try the latest GIT, I reverted the last patch and tried to solve the > problem differently. > > > On Tue, Feb 1, 2011 at 2:11 PM, Dan Lane <null@invalid.name> wrote: >> Yes, using 1f1541b our calls that use mod_loopback into mod_conference >> are unusable. The audio is choppy and delay increases as time passes. >> >> I thought it was an issue relating to the default 100Hz kernel on EC2 >> so I spent some time yesterday putting together a 1000Hz kernel but it >> didn't make any difference. In the meantime I've compiled mod_loopback >> from 4c5426f and loaded it with my 1f1541b build which eliminates the >> issue. >> >> I haven't added it to Jira yet as I want to spend some time debugging >> it (and I also owe you some debug info for FS-2934) but the problem is >> definitely there. >> >> On Tue, Feb 1, 2011 at 3:58 PM, Anthony Minessale >> <anthony.minessale@gmail.com> wrote: >>> Are you saying you have better results on that version than you do on >>> the latest? >>> What conditions do you have that cause you trouble, what is the >>> endpoint on the other side. >>> >>> If the last commit to mod_loopback intended to improve audio quality >>> actually makes it worse I need to investigate it. >>> >>> >>> On Tue, Feb 1, 2011 at 5:53 AM, Dan Lane <null@invalid.name> wrote: >>>> FWIW we've been also been having audio issues with loopback recently >>>> on EC2 (with a 1000Hz kernel). >>>> >>>> We worked around it in the short term by reverting mod_loopback to >>>> git-4c5426f during the build process. >>>> >>>> For anyone else who wants to try this just run "git checkout 4c5426f" >>>> in src/mod/endpoints/mod_loopback then build as usual. This is NOT a >>>> long term solution though. >>>> >>>> On Mon, Jan 31, 2011 at 8:02 AM, Øyvind Albrigtsen <oa@estation.dk> wrote: >>>>> Thanks for all your feedback. I'll keep on trying and inform you what >>>>> worked for me. >>>>> >>>>> >>>>> Regards >>>>> Oyvind >>>>> >>>>> On 2011-01-29 21:48, Anthony Minessale wrote: >>>>>> Everyone should try latest GIT before pondering any further because I >>>>>> added a patch like 2 days ago to adress this issue. >>>>>> >>>>>> >>>>>> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank@telonium.com> wrote: >>>>>> >>>>>>> Yeah. I, too, don't see the correlation between the NTP and hw timer.. >>>>>>> I am not familiar with the timer_test command and what it's measuring, but >>>>>>> of the 50 tests it ran, min is 19089 and max is 20713. >>>>>>> Frank >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone@ipeva.fr> >>>>>>> wrote: >>>>>>> >>>>>>>> Frank, >>>>>>>> I fail to see the relationship between the hw timer and NTP. >>>>>>>> Can you please elaborate ? >>>>>>>> David Ponzone Direction Technique >>>>>>>> email: david.ponzone@ipeva.fr >>>>>>>> tel: 01 74 03 18 97 >>>>>>>> gsm: 06 66 98 76 34 >>>>>>>> Service Client IPeva >>>>>>>> tel: 0811 46 26 26 >>>>>>>> www.ipeva.fr - www.ipeva-studio.com >>>>>>>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>>>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>>>>> non autorisée est interdite. Tout message électronique est susceptible >>>>>>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>>>>>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>>>>>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Le 28/01/2011 à 19:49, Frank Carmickle a écrit : >>>>>>>> >>>>>>>> Hi >>>>>>>> >>>>>>>> On Jan 28, 2011, at 7:39 AM, Øyvind Albrigtsen wrote: >>>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I'm using latest git-version of Freeswitch, and when I go to voicemail >>>>>>>> >>>>>>>> when calling a number the sound playback is choppy and it skips some of >>>>>>>> >>>>>>>> the digits in the number I called. >>>>>>>> >>>>>>>> What kind of results do you get from timer_test at the fs_cli? Are you >>>>>>>> running on hardware or are you virtualized? What is your clock source set >>>>>>>> to and what are your available clock source options? See >>>>>>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and >>>>>>>> /sys/devices/system/clocksource/clocksource0/current_clocksource. I am >>>>>>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to >>>>>>>> hang at 19998/19999 which works very well for me. When I was having problem >>>>>>>> it was reporting numbers all over the map from 17400 to 22600 with lots of >>>>>>>> randomness in between. I have my clocksource set to jiffies and xen >>>>>>>> independent wallclock set to 1. Of course at that point you need to have >>>>>>>> ntp running against a bunch of servers to drive your clock nice and steady. >>>>>>>> I know my set up is probably a lot different than yours but I thought I'd >>>>>>>> toss it out there to show that some of the harshest conditions can be dealt >>>>>>>> with and don't give up trying. If you are running on hardware with a cpu >>>>>>>> that doesn't have constant_tsc then you might have some problems. Just play >>>>>>>> with the different timer options until you find the one that works. >>>>>>>> >>>>>>>> HTH >>>>>>>> --FC >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> ----=======================---- >>>>>>> Frank Park >>>>>>> Telonium Communications, LLC >>>>>>> frank@telonium.com >>>>>>> http://www.telonium.com >>>>>>> Follow Us on Twitter: @GetTelonium >>>>>>> 404-566-8888 x1001 Office >>>>>>> 404-939-4242 Cell >>>>>>> ----=======================---- >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale@hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888@conference.freeswitch.org >>> googletalk:conf+888@conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best regards,
Dmitry Sytchev, IT Engineer
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (8)
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Anthony Minessale
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Dan Lane
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David Ponzone
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Dmitry Sytchev
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Frank Carmickle
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Frank Park
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Jeff Lenk
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Øyvind Albrigtsen