audo sync issues with record_session to mp3
Hi, We are running a cluster of 3 SIP servers all running Freeswitch. Each call is recorded using the record_session application to record to mp3 files using mod_shout. Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg. We are running record_session on the A-leg, and it seems to get ahead of the B-leg. For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. I am aware that recording to .mp3 can use additional resources, but these are quad core servers with 4GB of ram and they are only doing around 10 concurrent calls each. Additionally I have seen others experiencing these issues with Wav files too: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060107.html I am using the git head version from Tue 11 Oct 2011. I have also tried turning off enable_file_write_buffering on the A-leg however this makes no difference either. Does anyone have any suggestions? Thanks
On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott <tomp@tomp.co.uk> wrote:
Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg.
We are running record_session on the A-leg, and it seems to get ahead of the B-leg.
For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it.
What's on the other end of each leg? That might help us figure this out. Gabe
We have the same problem. We are running git from 12/30/11. our aleg is a sip channel coming to FS and the bleg is a sip channels leaving FS. I noticed this problem really when we started using mp3 instead of wav. With wav, it really was not noticeable for us in a 10-15minute call. But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so far out of sync, it sounds like 2 different calls. The relevant dialplan is <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Regex/XML Dialplan"> <context name="default"> <extension name="myextension"> <condition> <action application="sched_broadcast" data="+15740 sounds/announce.wav bleg"/> <action application="sched_hangup" data="+1800 "/> <action application="enum" data="15035411236"/> <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/mnt/rd/recordfile.mp3"/> <action application="bridge" data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: Thursday, October 20, 2011 3:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3 On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott <tomp@tomp.co.uk> wrote:
Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg.
We are running record_session on the A-leg, and it seems to get ahead of the B-leg.
For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it.
What's on the other end of each leg? That might help us figure this out. Gabe 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
I've just enabled session recording using MP3 encoding and have the same symptoms. In my case it is between a BRI ISDN connection and a SIP phone, both running with G.711. I haven't tried recording between extensions which would be a purer test, nor have I attempted using WAV. Are there any suggestions for how to fix this? Should I register a bug on Jira? John On 02/01/12 15:01, Frank @ Impact wrote:
We have the same problem. We are running git from 12/30/11. our aleg is a sip channel coming to FS and the bleg is a sip channels leaving FS.
I noticed this problem really when we started using mp3 instead of wav. With wav, it really was not noticeable for us in a 10-15minute call. But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so far out of sync, it sounds like 2 different calls.
The relevant dialplan is
<?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Regex/XML Dialplan"> <context name="default"> <extension name="myextension"> <condition> <action application="sched_broadcast" data="+15740 sounds/announce.wav bleg"/> <action application="sched_hangup" data="+1800 "/> <action application="enum" data="15035411236"/> <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/mnt/rd/recordfile.mp3"/> <action application="bridge" data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: Thursday, October 20, 2011 3:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3
On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott<tomp@tomp.co.uk> wrote:
Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg.
We are running record_session on the A-leg, and it seems to get ahead of the B-leg.
For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. What's on the other end of each leg? That might help us figure this out.
Gabe
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
I really wish things like this would go to jira it's very hard to track things from the mailing list...... commit a365fb636ad9e2f4bb5dee43eddc305560699114 Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Jan 11 17:49:35 2012 -0600 mailing list 36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk[Freeswitch-users] audo sync issues with record_session to mp3 I redid the stream recording with timestamps and headers to try to keep it more synced On Tue, Jan 10, 2012 at 8:46 AM, John <freeswitch@earthspike.net> wrote:
I've just enabled session recording using MP3 encoding and have the same symptoms. In my case it is between a BRI ISDN connection and a SIP phone, both running with G.711. I haven't tried recording between extensions which would be a purer test, nor have I attempted using WAV. Are there any suggestions for how to fix this? Should I register a bug on Jira?
John
On 02/01/12 15:01, Frank @ Impact wrote:
We have the same problem. We are running git from 12/30/11. our aleg is a sip channel coming to FS and the bleg is a sip channels leaving FS.
I noticed this problem really when we started using mp3 instead of wav. With wav, it really was not noticeable for us in a 10-15minute call. But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so far out of sync, it sounds like 2 different calls.
The relevant dialplan is
<?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Regex/XML Dialplan"> <context name="default"> <extension name="myextension"> <condition> <action application="sched_broadcast" data="+15740 sounds/announce.wav bleg"/> <action application="sched_hangup" data="+1800 "/> <action application="enum" data="15035411236"/> <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/mnt/rd/recordfile.mp3"/> <action application="bridge" data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: Thursday, October 20, 2011 3:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3
On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott<tomp@tomp.co.uk> wrote:
Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg.
We are running record_session on the A-leg, and it seems to get ahead of the B-leg.
For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. What's on the other end of each leg? That might help us figure this out.
Gabe
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
i did this before by recording in WAV then converted them to OGG - all WAV files of the day - after midnight where the usage is low. On Thu, Jan 12, 2012 at 9:00 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I really wish things like this would go to jira it's very hard to track things from the mailing list......
commit a365fb636ad9e2f4bb5dee43eddc305560699114 Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Jan 11 17:49:35 2012 -0600
mailing list 36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk[Freeswitch-users] audo sync issues with record_session to mp3 I redid the stream recording with timestamps and headers to try to keep it more synced
On Tue, Jan 10, 2012 at 8:46 AM, John <freeswitch@earthspike.net> wrote:
I've just enabled session recording using MP3 encoding and have the same symptoms. In my case it is between a BRI ISDN connection and a SIP phone, both running with G.711. I haven't tried recording between extensions which would be a purer test, nor have I attempted using WAV. Are there any suggestions for how to fix this? Should I register a bug on Jira?
John
On 02/01/12 15:01, Frank @ Impact wrote:
We have the same problem. We are running git from 12/30/11. our aleg is a sip channel coming to FS and the bleg is a sip channels leaving FS.
I noticed this problem really when we started using mp3 instead of wav. With wav, it really was not noticeable for us in a 10-15minute call. But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so far out of sync, it sounds like 2 different calls.
The relevant dialplan is
<?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Regex/XML Dialplan"> <context name="default"> <extension name="myextension"> <condition> <action application="sched_broadcast" data="+15740 sounds/announce.wav bleg"/> <action application="sched_hangup" data="+1800 "/> <action application="enum" data="15035411236"/> <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/mnt/rd/recordfile.mp3"/> <action application="bridge" data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: Thursday, October 20, 2011 3:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3
On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott<tomp@tomp.co.uk> wrote:
Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg.
We are running record_session on the A-leg, and it seems to get ahead of the B-leg.
For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. What's on the other end of each leg? That might help us figure this out.
Gabe
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
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
I ran a test this evening with a 15 minute phone call from PSTN via ISDN BRI (FreeTDM) into a SIP handset, all at G.711, recorded to MP3. Previously, this was out of sync. Now it's rock solid in sync. Thanks, Anthony! On 12/01/12 01:00, Anthony Minessale wrote:
I really wish things like this would go to jira it's very hard to track things from the mailing list......
commit a365fb636ad9e2f4bb5dee43eddc305560699114 Author: Anthony Minessale <anthm@freeswitch.org <mailto:anthm@freeswitch.org>> Date: Wed Jan 11 17:49:35 2012 -0600
mailing list 36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk <mailto:36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk> [Freeswitch-users] audo sync issues with record_session to mp3 I redid the stream recording with timestamps and headers to try to keep it more synced
On Tue, Jan 10, 2012 at 8:46 AM, John <freeswitch@earthspike.net <mailto:freeswitch@earthspike.net>> wrote:
I've just enabled session recording using MP3 encoding and have the same symptoms. In my case it is between a BRI ISDN connection and a SIP phone, both running with G.711. I haven't tried recording between extensions which would be a purer test, nor have I attempted using WAV. Are there any suggestions for how to fix this? Should I register a bug on Jira?
John
On 02/01/12 15:01, Frank @ Impact wrote: > We have the same problem. We are running git from 12/30/11. our aleg is a > sip channel coming to FS and the bleg is a sip channels leaving FS. > > I noticed this problem really when we started using mp3 instead of > wav. With wav, it really was not noticeable for us in a 10-15minute call. > But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so > far out of sync, it sounds like 2 different calls. > > The relevant dialplan is > > <?xml version="1.0"?> > <document type="freeswitch/xml"> > <section name="dialplan" description="Regex/XML Dialplan"> > <context name="default"> > <extension name="myextension"> > <condition> > <action application="sched_broadcast" data="+15740 > sounds/announce.wav bleg"/> > <action application="sched_hangup" data="+1800 "/> > <action application="enum" data="15035411236 <tel:15035411236>"/> > <action application="set" data="RECORD_ANSWER_REQ=true"/> > <action application="set" data="RECORD_STEREO=true"/> > <action application="record_session" > data="/mnt/rd/recordfile.mp3"/> > <action application="bridge" > data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> > <action application="hangup"/> > </condition> > </extension> > </context> > </section> > </document> > > > > -----Original Message----- > From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Gabriel > Gunderson > Sent: Thursday, October 20, 2011 3:07 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3 > > On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott<tomp@tomp.co.uk <mailto:tomp@tomp.co.uk>> wrote: >> Longer calls, after about 10 minutes start to introduce sync issues >> between the A-leg and the B-leg. >> >> We are running record_session on the A-leg, and it seems to get ahead of >> the B-leg. >> >> For example the caller on the A-leg will be heard to answer a question >> whilst the person on the B-leg is asking it. > What's on the other end of each leg? That might help us figure this out. > > Gabe > > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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 > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
participants (6)
-
Anthony Minessale
-
Frank ＠ Impact
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Gabriel Gunderson
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John
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Nandy Dagondon
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Tom Parrott