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

[Brian West]

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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>
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-- 

*Brian West*
brian at freeswitch.org


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