Re: [Freeswitch-users] Issues with rtp audio going in slow motion
Is this an issue with transcoding? I am wondering how I lock it down so that transcoding only happens at a minimum Also, would it be bad to completely remove G722 from the codec list? I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722 On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
Joel is there any way you can test with one polycom set to g711u. This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible) On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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I will look into that Chris, thank you. I did set the inbound codec to force g711a + u and also did the same on outbound calls heading to our ITSP. On Tue, Oct 28, 2014 at 7:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation? On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences. The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension. On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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--
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*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit) Current version on the server On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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--
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*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions. On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:
> I am having an issue where some calls start out fine and then go > into slow motion (distorted slow speech) > > I noticed something in the logs stating > > Activating write resampler > > > What does this mean and how do I avoid this in the future? > > > > Thank you in advance >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler. Any suggestions? I have not set the phone to only use G711u yet, I am trying to make this work on server side only. On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions.
On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:
> Is this an issue with transcoding? > > I am wondering how I lock it down so that transcoding only happens > at a minimum > > Also, would it be bad to completely remove G722 from the codec list? > > I think the issue is that our provider is sending G711 and all of > the Polycom Phones are trying to use G722 > > On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> > wrote: > >> I am having an issue where some calls start out fine and then go >> into slow motion (distorted slow speech) >> >> I noticed something in the logs stating >> >> Activating write resampler >> >> >> What does this mean and how do I avoid this in the future? >> >> >> >> Thank you in advance >> > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute? On Mon, Dec 15, 2014 at 12:14 PM, Joel White <joelewhite@gmail.com> wrote:
Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler.
Any suggestions?
I have not set the phone to only use G711u yet, I am trying to make this work on server side only.
On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions.
On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
> Also, just to mention. The CPU is never above 2% and the current > load is about 20 simultaneous calls > > On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> > wrote: > >> Is this an issue with transcoding? >> >> I am wondering how I lock it down so that transcoding only happens >> at a minimum >> >> Also, would it be bad to completely remove G722 from the codec list? >> >> I think the issue is that our provider is sending G711 and all of >> the Polycom Phones are trying to use G722 >> >> On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> >> wrote: >> >>> I am having an issue where some calls start out fine and then go >>> into slow motion (distorted slow speech) >>> >>> I noticed something in the logs stating >>> >>> Activating write resampler >>> >>> >>> What does this mean and how do I avoid this in the future? >>> >>> >>> >>> Thank you in advance >>> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
To file a jira, just go to freeswitch.org/jira and follow the signup link it takes about a minute, once you have you account, you can then open tickets... See https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA for more details on filing the jira On 12/15/14 11:15 AM, "Joel White" <joelewhite@gmail.com> wrote:
Anthony, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute?
On Mon, Dec 15, 2014 at 12:14 PM, Joel White <joelewhite@gmail.com> wrote:
Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler.
Any suggestions?
I have not set the phone to only use G711u yet, I am trying to make this work on server side only.
On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions.
On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote: > Joel is there any way you can test with one polycom set to g711u. > > This sounds to me like a 16k vs 8k transcoding issue, maybe setup your > codecs on the freeswitch side to only allow 8k g722 (if that's even > possible) > > On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> > wrote: Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote: Is this an issue with transcoding? I am wondering how I lock it down so that transcoding only happens at a minimum Also, would it be bad to completely remove G722 from the codec list? I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722 On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote: I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech) I noticed something in the logs stating Activating write resampler What does this mean and how do I avoid this in the future? Thank you in advance _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
Will do, thank you for the information. On Mon, Dec 15, 2014 at 12:19 PM, Ken Rice <krice@freeswitch.org> wrote:
To file a jira, just go to freeswitch.org/jira and follow the signup link it takes about a minute, once you have you account, you can then open tickets...
See https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA for more details on filing the jira
On 12/15/14 11:15 AM, "Joel White" <joelewhite@gmail.com> wrote:
Anthony, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute?
On Mon, Dec 15, 2014 at 12:14 PM, Joel White <joelewhite@gmail.com> wrote:
Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler.
Any suggestions?
I have not set the phone to only use G711u yet, I am trying to make this work on server side only.
On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions.
On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Joel is there any way you can test with one polycom set to g711u.
This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)
On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:
Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls
On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote: Is this an issue with transcoding?
I am wondering how I lock it down so that transcoding only happens at a minimum
Also, would it be bad to completely remove G722 from the codec list?
I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722
On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote: I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)
I noticed something in the logs stating
Activating write resampler
What does this mean and how do I avoid this in the future?
Thank you in advance
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
You sign up for an account, log in, create an issue and fill out the fields. Make sure you have reproduced your problem on the very latest master from git before reporting because its a mandatory question. On Mon, Dec 15, 2014 at 11:15 AM, Joel White <joelewhite@gmail.com> wrote:
Anthony, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute?
On Mon, Dec 15, 2014 at 12:14 PM, Joel White <joelewhite@gmail.com> wrote:
Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler.
Any suggestions?
I have not set the phone to only use G711u yet, I am trying to make this work on server side only.
On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions.
On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com > wrote:
> Joel is there any way you can test with one polycom set to g711u. > > This sounds to me like a 16k vs 8k transcoding issue, maybe setup > your codecs on the freeswitch side to only allow 8k g722 (if that's even > possible) > > On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> > wrote: > >> Also, just to mention. The CPU is never above 2% and the current >> load is about 20 simultaneous calls >> >> On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> >> wrote: >> >>> Is this an issue with transcoding? >>> >>> I am wondering how I lock it down so that transcoding only happens >>> at a minimum >>> >>> Also, would it be bad to completely remove G722 from the codec >>> list? >>> >>> I think the issue is that our provider is sending G711 and all of >>> the Polycom Phones are trying to use G722 >>> >>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com >>> > wrote: >>> >>>> I am having an issue where some calls start out fine and then go >>>> into slow motion (distorted slow speech) >>>> >>>> I noticed something in the logs stating >>>> >>>> Activating write resampler >>>> >>>> >>>> What does this mean and how do I avoid this in the future? >>>> >>>> >>>> >>>> Thank you in advance >>>> >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.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 > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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 >
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
2014-12-15 10:13:41.728748 [DEBUG] switch_core_media.c:5083 Audio params are unchanged for sofia/internal/51024@voip.net. 2014-12-15 10:13:41.748724 [DEBUG] switch_core_media.c:2381 Changing Codec from PCMU@20ms@8000hz to G722@20ms@8000hz 2014-12-15 10:13:41.768769 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/14073134300@69.174.248.123 [BREAK] 2014-12-15 10:13:41.788777 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/51024@bcvoip.net G722/8000 20 ms 160 samples 64000 bits 1 channels 2014-12-15 10:13:41.788777 [DEBUG] switch_core_codec.c:123 sofia/internal/ 51024@voip.net Original read codec replaced with G722:9 2014-12-15 10:13:41.808784 [NOTICE] switch_core_io.c:1287 Activating write resampler 2014-12-15 10:13:41.808784 [DEBUG] switch_core_media.c:2055 alternate payload received (received 0, expecting 9) 2014-12-15 10:13:41.808784 [WARNING] switch_core_media.c:2066 Changing current codec to PCMU (payload type 0). 2014-12-15 10:13:41.808784 [NOTICE] switch_core_io.c:1287 Activating write resampler 2014-12-15 10:13:41.828776 [NOTICE] switch_core_media.c:2374 Deactivating write resampler 2014-12-15 10:13:41.828776 [DEBUG] switch_core_media.c:2381 Changing Codec from G722@20ms@8000hz to PCMU@20ms@8000hz 2014-12-15 10:13:41.848763 [NOTICE] switch_core_io.c:1287 Activating write resampler 2014-12-15 10:13:41.868765 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/51024@voip.net PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2014-12-15 10:13:41.868765 [DEBUG] switch_core_codec.c:123 sofia/internal/ 51024@voip.net Original read codec replaced with PCMU:0 Some of the messages I see that concern me about the call On Mon, Dec 15, 2014 at 12:15 PM, Joel White <joelewhite@gmail.com> wrote:
Anthony, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute?
On Mon, Dec 15, 2014 at 12:14 PM, Joel White <joelewhite@gmail.com> wrote:
Interesting information I found out about this issue. One person stated that they had an hour long call. About mid way through the call the caller dropped down to this "slow motion" slower sample rate. After about 30 seconds of this slow motion, the call went back to a normal state and worked fine for the rest of the call. I am forcing G711u on all inbound and outbound calls heading the to PSTN / VoIP carrier. This is an interesting question and makes me wonder if I can disable the write resampler.
Any suggestions?
I have not set the phone to only use G711u yet, I am trying to make this work on server side only.
On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions.
On Mon, Nov 3, 2014 at 10:49 AM, Joel White <joelewhite@gmail.com> wrote:
FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)
Current version on the server
On Mon, Nov 3, 2014 at 11:44 AM, Joel White <joelewhite@gmail.com> wrote:
I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences.
The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.
On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:
What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?
On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com > wrote:
> Joel is there any way you can test with one polycom set to g711u. > > This sounds to me like a 16k vs 8k transcoding issue, maybe setup > your codecs on the freeswitch side to only allow 8k g722 (if that's even > possible) > > On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> > wrote: > >> Also, just to mention. The CPU is never above 2% and the current >> load is about 20 simultaneous calls >> >> On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> >> wrote: >> >>> Is this an issue with transcoding? >>> >>> I am wondering how I lock it down so that transcoding only happens >>> at a minimum >>> >>> Also, would it be bad to completely remove G722 from the codec >>> list? >>> >>> I think the issue is that our provider is sending G711 and all of >>> the Polycom Phones are trying to use G722 >>> >>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com >>> > wrote: >>> >>>> I am having an issue where some calls start out fine and then go >>>> into slow motion (distorted slow speech) >>>> >>>> I noticed something in the logs stating >>>> >>>> Activating write resampler >>>> >>>> >>>> What does this mean and how do I avoid this in the future? >>>> >>>> >>>> >>>> Thank you in advance >>>> >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.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 > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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 >
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 (5)
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Anthony Minessale
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Brian West
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Chris Tunbridge
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Joel White
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Ken Rice