[Freeswitch-users] Issues with rtp audio going in slow motion

[Joel White]

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 at 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 at 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 at 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 at 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 at 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 at 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
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>
>>>
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>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
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>>
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>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
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>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>
>
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