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

[Joel White]

Will do, thank you for the information.

On Mon, Dec 15, 2014 at 12:19 PM, Ken Rice <krice at 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 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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