<p>fsctl debug_level 10<br>
Also make sure ve host has high res timing.</p>
<div class="gmail_quote">On Jun 24, 2013 8:06 PM, "Nathan Neulinger" <<a href="mailto:nneul@mst.edu">nneul@mst.edu</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
This instance is running v1.2stable as of a week or so ago. (In a pilot, so trying not to update it very frequently, but<br>
when I add stuff to the skinny code.)<br>
<br>
The interesting thing is - on MOST calls, it's working fine, and even on the calls that are a problem, it works fine, up<br>
until "something" happens, and then it starts building up the delay.<br>
<br>
Even on non-problem calls, I occasionally see a slow/occasional stairstep that doesn't accumulate much, but on the BAD<br>
calls it accumulates to high delays.<br>
<br>
Example call - graphs from voipmonitor:<br>
<br>
<a href="http://imgur.com/F4S91mr,lAulG2o" target="_blank">http://imgur.com/F4S91mr,lAulG2o</a><br>
<br>
-- Nathan<br>
<br>
On 06/24/2013 07:38 PM, William King wrote:<br>
> I've seen a similar issue on a small network behind a pfsense router.<br>
> Network diagram is:<br>
> Hard Phones(3+ Yealink and Polycom mix) -> switch -> PFsense -> Comcast<br>
> -> FS conferencing server.<br>
><br>
> In my case I was seeing ~10 seconds of RTP delay(tested from doing a/b<br>
> response testing), but instant sip handling. The audio was in perfect<br>
> sync coming from the conference server to the 3 endpoints.<br>
><br>
> Nathan, what version of FS are you running?<br>
><br>
> I have not yet ruled out that the issue isn't a local network problem.<br>
><br>
> William King<br>
> Senior Engineer<br>
> Quentus Technologies, INC<br>
> 1037 NE 65th St Suite 273<br>
> Seattle, WA 98115<br>
> Main: <a href="tel:%28877%29%20211-9337" value="+18772119337">(877) 211-9337</a><br>
> Office: <a href="tel:%28206%29%20388-4772" value="+12063884772">(206) 388-4772</a><br>
> Cell: <a href="tel:%28253%29%20686-5518" value="+12536865518">(253) 686-5518</a><br>
> <a href="mailto:william.king@quentustech.com">william.king@quentustech.com</a><br>
><br>
> On 06/24/2013 04:40 PM, Nathan Neulinger wrote:<br>
>> In one particular stream, which is call mgr <-> FS, the CCM to freeswitch side of the stream shows no issues, but<br>
>> freeswitch to CCM direction shows a number of drifts, some just one ptime worth, others larger.<br>
>><br>
>> This instance is running as a VM, but allocated significantly more reserved cpu (and very high shares) on a lightly<br>
>> loaded host. Concurrent calls on the FS instance is in the single digits at most at this point. Typical end to end<br>
>> network induced latencies less than 3 ms at most.<br>
>><br>
>> Any suggestions on what to look at or steps to take here? Certainly not happening on every call, but I do believe I've<br>
>> seen it on both cisco and polycom phone connections. Usually, there is just a slow stairstep over the period of the call.<br>
>><br>
>> -- Nathan<br>
>><br>
>> On 06/24/2013 06:24 PM, Nathan Neulinger wrote:<br>
>>> Having some occasional user reports of problem calls where they are fine, and then something happens on the call and lag<br>
>>> increases significantly, sometimes up to 2+ seconds.<br>
>>><br>
>>> I've got severe examples on both cisco and polycom phones. What I'm seeing looks like the call is going perfectly for a<br>
>>> period of time (sometimes up to a few minutes), and then there is a period of time with higher latency packets, and the<br>
>>> overall delay in the graph increases steadily at that point, and never recovers.<br>
>>><br>
>>> It sounded like rtp-autoflush-during-bridge might be something relevant, but that's set to the default. Also saw<br>
>>> "rtp-rewrite-timestamps", but unclear if that is necessary here.<br>
>>><br>
>>> I've got pcap captures of the rtp streams if that'll help any.<br>
>>><br>
>>> -- Nathan<br>
>>><br>
>>> ------------------------------------------------------------<br>
>>> Nathan Neulinger <a href="mailto:nneul@mst.edu">nneul@mst.edu</a><br>
>>> Missouri S&T Information Technology <a href="tel:%28573%29%20612-1412" value="+15736121412">(573) 612-1412</a><br>
>>> System Administrator - Architect<br>
>>><br>
>>> _________________________________________________________________________<br>
>>> Professional FreeSWITCH Consulting Services:<br>
>>> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>>><br>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>>><br>
>>> Official FreeSWITCH Sites<br>
>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
>>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
>>><br>
>>> FreeSWITCH-users mailing list<br>
>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
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>>><br>
>><br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
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><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
><br>
> Official FreeSWITCH Sites<br>
> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
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><br>
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><br>
<br>
--<br>
------------------------------------------------------------<br>
Nathan Neulinger <a href="mailto:nneul@mst.edu">nneul@mst.edu</a><br>
Missouri S&T Information Technology <a href="tel:%28573%29%20612-1412" value="+15736121412">(573) 612-1412</a><br>
System Administrator - Architect<br>
<br>
_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
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<br>
FreeSWITCH-users mailing list<br>
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</blockquote></div>