Summary:ASTERISK-15952: sip show channelstats isses.
Reporter:Balint Cristian (cbalint)Labels:
Date Opened:2010-04-14 00:00:22Date Closed:2011-06-07 14:04:47
Versions:Frequency of
Description:- Product version, Fedora 12.

- "sip show channelstats" not always measure RX/TX packets.

When it doesnt measure it displays like this (~3minute long call):
Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter      1d36e028-62  00:02:49 0000000000  0000000000 ( 0.00%) 000000 0000000000  0000000000 ( 0.00%) 000000      63cdd90f403  00:02:49 0000000000  0000000000 ( 0.00%) 000000 0000000000  0000000000 ( 0.00%) 000000
- In my observations first call is measured, than second,third and so on isn't.
- Also "Pack  Lost" sometimes has abberant values, but rx/tx would be enough for me,so ignore "loss" fields.

- Our scenario is:
SIP-proveder<---VLAN--->asterisk<--->[our network]

- We always use pass-through asterisk the rtp flow, never a call is re-invited.
- We use alaw, but same is with ulaw codec.
- Double checked using ethereal and calls are not redirected not even by accident.

- We use to log RX/TX packets to log in CDR too:
exten => h,n,Set(CDR(rxpackets)=${CHANNEL(rtpqos,audio,local_count)})
exten => h,n,Set(CDR(txpackets)=${CHANNEL(rtpqos,audio,remote_count)})
- But same results are with "sip show channelstats" too.


- Its very important for us to see per-call rx/tx statistics to be able see if those values are highly assymetric (e.g. 200 packets difference => 5 sec audio hang/blank) to point out those calls which are hanged or droped frecvently due to short network outages or packet losses, so be able to check customers line quality in form of statistics and avoid annoing "call was hanged again" from customer side. Unfortunate is when call was placed but no rx/tx record was registered, so we have no statistics at all.

I have good C/C++ programming skills but I will have hard time catch this issue down, especialy that I need a serious testbed and cannnot do that over a production enviroment.
Comments:By: Leif Madsen (lmadsen) 2010-04-14 09:53:30

You may want to try the PineFrog branch by oej:  http://www.voip-forum.com/opensource/2010-01/test-rtcp-test-branch-based-asterisk-14/

By: Leif Madsen (lmadsen) 2010-04-14 09:54:18

You probably also need to provide some debugging information such as a pcap trace and some console output that shows the data that is incorrect.

By: Paul Belanger (pabelanger) 2010-05-12 12:39:19

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 can be found at http://www.asterisk.org/developers/bug-guidelines