Summary: | ASTERISK-24973: Wacky Send: Pack Lost ( %) Jitter | ||
Reporter: | Christophe Prevotaux (cp) | Labels: | |
Date Opened: | 2015-04-17 11:36:38 | Date Closed: | 2015-06-16 14:43:55 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | 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 14.04 | Attachments: | ( 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 |