[asterisk-bugs] [Asterisk 0017177]: sip show channelstats isses.
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Apr 14 09:53:30 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17177
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Reported By: cbalint
Assigned To:
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Project: Asterisk
Issue ID: 17177
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
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:
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Date Submitted: 2010-04-14 00:00 CDT
Last Modified: 2010-04-14 09:53 CDT
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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.
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Relationships ID Summary
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related to 0016280 RTPAUDIOQOS and RTPAUDIOQOSBRIDGED fals...
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(0120396) lmadsen (administrator) - 2010-04-14 09:53
https://issues.asterisk.org/view.php?id=17177#c120396
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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/
Issue History
Date Modified Username Field Change
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2010-04-14 09:53 lmadsen Note Added: 0120396
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