[asterisk-bugs] [Asterisk 0017177]: sip show channelstats isses.

Asterisk Bug Tracker noreply at bugs.digium.com
Wed May 12 12:39:19 CDT 2010


The following issue has been CLOSED 
====================================================================== 
https://issues.asterisk.org/view.php?id=17177 
====================================================================== 
Reported By:                cbalint
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17177
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.6 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             2010-04-14 00:00 CDT
Last Modified:              2010-05-12 12:39 CDT
====================================================================== 
Summary:                    sip show channelstats isses.
Description: 
- Product version 1.6.2.6-i686, 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
84.1.226.38      1d36e028-62  00:02:49 0000000000  0000000000 ( 0.00%)
000000 0000000000  0000000000 ( 0.00%) 000000
10.1.163.12      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.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0016280 RTPAUDIOQOS and RTPAUDIOQOSBRIDGED fals...
====================================================================== 

---------------------------------------------------------------------- 
 (0121807) pabelanger (manager) - 2010-05-12 12:39
 https://issues.asterisk.org/view.php?id=17177#c121807 
---------------------------------------------------------------------- 
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 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-05-12 12:39 pabelanger     Note Added: 0121807                          
2010-05-12 12:39 pabelanger     Status                   feedback => closed  
======================================================================




More information about the asterisk-bugs mailing list