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

[Ken Rice]

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On 12/15/14 11:15 AM, "Joel White" <joelewhite at 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 at 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 at 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 at 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 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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-- 
Ken
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