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Summary:ASTERISK-24973: Wacky Send: Pack Lost ( %) Jitter
Reporter:Christophe Prevotaux (cp)Labels:
Date Opened:2015-04-17 11:36:38Date Closed:2015-06-16 14:43:55
Priority:MinorRegression?
Status:Closed/CompleteComponents:Core/General
Versions:11.17.0 11.17.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment:FreeBSD 10.1 , FreePBX ( Asterisk 11.17.1 built by root @ jenkins-builder1.schmoozecom.net on a i686 running Linux on 2015-04-14 20:59:38 UTC), Ubuntu 14.04Attachments:( 0) asterisk-config.tar.gz
( 1) full.bz2
Description:Happens after running traffic for more than 52 minutes on my machines


{noformat}
localhost*CLI> sip show channelstats
Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
192.168.0.151    43a37f3440c  01:49:14 0000000328K 0000000008 ( 0.00%) 3.0000 0000000327K 0000000000 ( 0.00%) 0.0037
192.168.0.149    0cbc65b6551  01:49:06 0000000327K 0000000015 ( 0.00%) 3.0000 0000000327K 0000000000 ( 0.00%) 0.0048
192.168.0.157    7b24421a5e4  01:49:24 0000000328K 0000000011 ( 0.00%) 1.0000 0000000328K 0000000000 ( 0.00%) 0.0034
192.168.0.155    7dbd60e4-22  01:49:00 0000000327K 0000000059 ( 0.02%) 1.0000 0000000327K 0000000000 ( 0.00%) 0.0034
192.168.0.148    f8815589-20  01:49:10 0000000327K 0000000007 ( 0.00%) 2.0000 0000000327K 0016777215 (5116.83%) 0.0029
192.168.0.158    f7cb3fac-23  01:49:19 0000000328K 0000000013 ( 0.00%) 2.0000 0000000328K 0000000002 ( 0.00%) 0.0042
192.168.0.156    5bcb7b0f-21  01:49:06 0000000327K 0000000019 ( 0.01%) 2.0000 0000000327K 0000000000 ( 0.00%) 0.0032
192.168.0.152    6c3e594c167  01:49:00 0000000327K 0000000008 ( 0.00%) 2.0000 0000000327K 0016777215 (5127.51%) 0.0030
192.168.0.153    63792575-1f  01:49:24 0000000328K 0000000011 ( 0.00%) 2.0000 0000000328K 0000000000 ( 0.00%) 0.0046
192.168.0.150    fe08e964-22  01:49:14 0000000327K 0000000007 ( 0.00%) 1.0000 0000000328K 0000000000 ( 0.00%) 0.0044
192.168.0.154    728c80961ac  01:49:19 0000000328K 0000000004 ( 0.00%) 3.0000 0000000328K 0000000000 ( 0.00%) 0.0039
192.168.0.133    436b9fdf1f0  01:49:10 0000000327K 0000000007 ( 0.00%) 2.0000 0000000327K 0000000000 ( 0.00%) 0.0043
{noformat}
Comments:By: Matt Jordan (mjordan) 2015-04-17 12:25:42.243-0500

As I mentioned in #asterisk, please provide:

# Your relevant configuration files for the channels involved, including a {{sip.conf}}/{{pjsip.conf}} and {{extensions.conf}}
# Explicit instructions for reproducing the problem. "It happened on my machine" is not sufficient. We need to know how you are establishing the calls, whether or not the network for those calls traverses a NAT, etc.
# Log files for the message traffic involving the affected channels. This may include setting "rtcp set debug on" for those channels.

By: Eric Bree (ebree@nltinc.com) 2015-04-20 08:33:55.436-0500

tar of /etc/asterisk directory
By: Eric Bree (ebree@nltinc.com) 2015-04-20 08:45:55.379-0500

I have attached the config files.  The physical setup is the asterisk server, connected to a switch, an Access Point connected to the switch and 12 WiFi phones connected to the Access Point.  Each phone is configured with extensions 001 to 012 and a call is placed between pairs of phones (001 to 007, 002 to 008, etc..).

I have enabled "rtcp set debug on" this morning and will let it run for an hour or so before I upload the log file.
By: Eric Bree (ebree@nltinc.com) 2015-04-20 09:56:36.771-0500

I have uploaded the asterisk config files, along with the log file with "rtcp set debug on".
By: Eric Bree (ebree@nltinc.com) 2015-04-20 10:10:45.696-0500

New "sip show channelstats" to go with the uploaded logfile.

{code}
localhost*CLI> sip show channelstats
Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
192.168.0.153    7974f5c2-15  01:11:17 0000000213K 0000000016 ( 0.01%) 3.0000 0000000213K 0000000000 ( 0.00%) 0.0010
192.168.0.155    a5a34480-15  01:09:43 0000000209K 0000000002 ( 0.00%) 0.0000 0000000209K 0000000000 ( 0.00%) 0.0012
192.168.0.148    732535980fc  01:10:11 0000000210K 0000000006 ( 0.00%) 3.0000 0000000210K 0000000000 ( 0.00%) 0.0012
192.168.0.156    fa1ecad5-15  01:09:53 0000000209K 0000000062 ( 0.03%) 2.0000 0000000209K 0000000007 ( 0.00%) 0.0012
192.168.0.149    008fcd9428c  01:09:52 0000000209K 0000000003 ( 0.00%) 3.0000 0000000209K 0000000000 ( 0.00%) 0.0028
192.168.0.157    33acb154312  01:11:17 0000000213K 0000000012 ( 0.01%) 1.0000 0000000213K 0016777215 (7843.30%) 0.0010
192.168.0.158    50af1571-15  01:10:57 0000000212K 0000000003 ( 0.00%) 2.0000 0000000213K 0000000000 ( 0.00%) 0.0013
192.168.0.152    443989f8364  01:09:42 0000000209K 0000000006 ( 0.00%) 1.0000 0000000209K 0000000000 ( 0.00%) 0.0012
192.168.0.133    f47e3dc5-52  01:10:11 0000000210K 0000000007 ( 0.00%) 3.0000 0000000210K 0000000000 ( 0.00%) 0.0011
192.168.0.154    078982bf40a  01:10:57 0000000213K 0000000007 ( 0.00%) 0.0000 0000000212K 0000000001 ( 0.00%) 0.0010
192.168.0.151    365c0586446  01:10:34 0000000211K 0000000007 ( 0.00%) 1.0000 0000000211K 0000000000 ( 0.00%) 0.0011
192.168.0.150    a13b469b-15  01:10:34 0000000211K 0000000005 ( 0.00%) 3.0000 0000000211K 0000000000 ( 0.00%) 0.0011
12 active SIP channels
{code}
By: Rusty Newton (rnewton) 2015-05-03 08:35:01.304-0500

Is the problem only with the stats that show up? Or are there symptoms like no audio or garbled audio presented to the user?

You mentioned the problem happens after 52 minutes. Is it exactly 52 minutes every time?
By: Christophe Prevotaux (cp) 2015-05-11 08:36:12.934-0500

At this point we have to say that this was a test for connectivity and there was no one listening in on the calls. We just ran these tests to make sure calls would not drop in our setup.
So we can not tell whether or not this correlated with some garbled audio or no audio. For this we would have to run these tests again and listen in.
By: Rusty Newton (rnewton) 2015-06-01 17:10:47.889-0500

Thanks for the data. I have a few comments.

* The logfile is too large to be useful. It might be better to provide a log where it begins at the beginning of a suspect call and ends after it.
* Your 'sip show channelstats' command output shows a call with a Call ID that includes the string "33acb154312". I cannot find this call in your log file.
* You have provided all of your configuration files. This could be helpful if we had the same environment, hardware and software as you, but as it is we don't much to go on. We can't simply browse every single config file looking for abnormalities (which we are not going to do anyway since this isn't technical support).

Unfortunately the data is not useful so far.

If you can provided a narrower set of configuration files that will reproduce the issue, along with instructions on how to use those config files to reproduce the issue (e.g. Setup Phones A and B with the endpoints in pjsip.conf and then make a call between them..) then we could possibly make some progress.

I'm not sure what else we can do unless we get an idea on how to reproduce the issue. You could provide a packet capture of a suspect call (along with an Asterisk log of that call), but I'm not certain that will help. If you do this, you would want to have the *DEBUG* channel on in your log file so that we can all the information available for debugging.

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: Rusty Newton (rnewton) 2015-06-16 14:43:47.317-0500

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines