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<TITLE>Re: [Freeswitch-users] Issues with rtp audio going in slow motion</TITLE>
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<FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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...<BR>
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See <a href="https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA">https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA</a> for more details on filing the jira<BR>
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On 12/15/14 11:15 AM, "Joel White" <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Anthony, I am not signed up to JIRA and have never used it. Do you have any information I need to know to contribute?<BR>
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On Mon, Dec 15, 2014 at 12:14 PM, Joel White <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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.<BR>
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Any suggestions?<BR>
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I have not set the phone to only use G711u yet, I am trying to make this work on server side only.<BR>
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On Mon, Nov 3, 2014 at 12:00 PM, Anthony Minessale <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>This should be handled in JIRA not the mailing list.<BR>
If problem or issue seems prudent in the subject, think JIRA <a href="http://jira.freeswitch.org">http://jira.freeswitch.org</a><BR>
That way you can post logs etc and not flood the list with help questions.<BR>
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On Mon, Nov 3, 2014 at 10:49 AM, Joel White <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)<BR>
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Current version on the server<BR>
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On Mon, Nov 3, 2014 at 11:44 AM, Joel White <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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. <BR>
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The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.<BR>
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On Mon, Nov 3, 2014 at 10:42 AM, Brian West <<a href="brian@freeswitch.org">brian@freeswitch.org</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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?<BR>
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On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <<a href="blasterjr@gmail.com">blasterjr@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Joel is there any way you can test with one polycom set to g711u.<BR>
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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)<BR>
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On Tue, Oct 21, 2014 at 8:22 AM, Joel White <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
</SPAN></FONT></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls <BR>
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On Tue, Oct 21, 2014 at 10:16 AM, Joel White <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
Is this an issue with transcoding?<BR>
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I am wondering how I lock it down so that transcoding only happens at a minimum<BR>
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Also, would it be bad to completely remove G722 from the codec list?<BR>
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I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722<BR>
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On Mon, Oct 20, 2014 at 11:43 AM, Joel White <<a href="joelewhite@gmail.com">joelewhite@gmail.com</a>> wrote:<BR>
I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)<BR>
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I noticed something in the logs stating <BR>
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Activating write resampler<BR>
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What does this mean and how do I avoid this in the future?<BR>
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Thank you in advance <BR>
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