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

Joel White joelewhite at gmail.com
Mon Dec 15 20:21:50 MSK 2014


2014-12-15 10:13:41.728748 [DEBUG] switch_core_media.c:5083 Audio params
are unchanged for sofia/internal/51024 at voip.net.
2014-12-15 10:13:41.748724 [DEBUG] switch_core_media.c:2381 Changing Codec
from PCMU at 20ms@8000hz to G722 at 20ms@8000hz
2014-12-15 10:13:41.768769 [DEBUG] switch_core_session.c:1053 Send signal
sofia/external/14073134300 at 69.174.248.123 [BREAK]
2014-12-15 10:13:41.788777 [DEBUG] switch_core_media.c:2473 Set Codec
sofia/internal/51024 at 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 at 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 at 20ms@8000hz to PCMU at 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 at 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 at 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 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
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
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>>>>>>>> http://www.cluecon.com
>>>>>>>>
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>>>>>>>>
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>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
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>>>>>>>
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>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *Brian West*
>>>>>> brian at 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 at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
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>>>>>> http://www.cluecon.com
>>>>>>
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>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
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>>>> 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+
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>>>
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>>>
>>>
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>>>
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>>
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