[asterisk-users] sip show channelstats reliable?

Scott Griepentrog sgriepentrog at digium.com
Mon Jan 19 13:13:01 CST 2015


I would recommend capturing traffic outside your Asterisk server with
Wireshark, then running the Telephony/Rtp/Analysize Streams option to
determine if you have packet loss at that point in the network.

On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote:

> Thanks but no Adtran here.
>
> I do think these stats are indicating an issue, I just don't know how to
> prove it outside Asterisk.
>
>
> ------------------------------
> From: EWieling at nyigc.com
> To: tjrlist at live.com; asterisk-users at lists.digium.com
> Date: Mon, 19 Jan 2015 13:55:33 -0500
> Subject: RE: [asterisk-users] sip show channelstats reliable?
>
>
> I’ve seen something similar with Adtran SIP gateways.    When a re-invite
> happens the Adtran gets all confused about call stats and marks the
> pre-reinvite leg of the call as losing large numbers of packets.    BTW,
> IIRC reinvites happen when a codec changes or the channel switches to T.38.
>
>
>
> Also Adtran SIP gateways appear not to support OPTIONS packets when
> running in SIP proxy mode, which is very annoying.     At some point I’ll
> try and arrange a slugfest between Digium and Adtran and they can figure
> out why it doesn’t work.
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Todd R.
> *Sent:* Monday, January 19, 2015 1:45 PM
> *To:* Asterisk-Users List
> *Subject:* Re: [asterisk-users] sip show channelstats reliable?
>
>
>
> Additional info:
>
>
>
> At the moment I am running 1.8.x but the other day I was getting the same
> results on 11.x
>
>
>
> Here is a sample from show channelstats. I do think this command is
> showing that there is trouble between specific IP's and my Asterisk box but
> I don't know if the numbers are accurate and reliable.
>
>
>
> Peer
>
> Call ID
>
> Duration
>
> Recv: Pack
>
> Lost
>
> (     %)
>
> Jitter
>
> Send: Pack
>
> Lost
>
> (
>
> %)
>
> Jitter
>
> x.x.x.x
>
> 5531341d06b
>
> 00:07:42
>
> 0000023123
>
> 0000063836
>
> (73.41%)
>
> 0.0000
>
> 0000023102
>
> 0000000000
>
> (
>
> 0.00%)
>
> 0.0007
>
>
>
> Peer IP changed to protect the innocent :-)
>
>
> ------------------------------
>
> From: tjrlist at live.com
> To: asterisk-users at lists.digium.com
> Date: Mon, 19 Jan 2015 12:17:25 -0600
> Subject: [asterisk-users] sip show channelstats reliable?
>
> I am seeing lots of lost packets when running the command sip show
> channelstats at the CLI.
>
>
>
> There are issues across multiple Asterisk servers I am trying to diagnose
> but everything I read seems to point to this command being pretty
> unreliable.
>
>
>
> Can I trust the info this command shows?
>
>
>
> I am showing lots of lost packets in sip show channelstats but I can't see
> any packet loss when pinging the same IP's to/from.
>
>
>
> Since I don't 100% control the network my gear is on, I need something
> outside of Asterisk to show the network engineer to convince here and
> myself that there are network issues.
>
>
>
> All I have is the loss that's shown from this command with no real network
> stats to back it up.
>
>
>
> Is there a magic command in CentOS anyone can recommend to diagnose and
> match up the issues shown in Asterisk using this command?
>
>
>
> Moving gear around on the network changes the info Asterisk shows a LOT.
> For example, if I point traffic to the main physical gateway I get loss to
> a particular customer's IP (their PBX), if I move it to another place on
> the network (as a VM) their IP is good and other customers IP's start
> showing loss using the channelstats info.
>
>
>
> Driving me freakin' crazy. It does appear there are network issues causing
> my troubles but I can't get help if I can't point to some hard and fast
> issues outside of Asterisk.
>
>
>
> The only thing I have right now is collissions showing on one of a few of
> our pfSense devices but they are virtual running on XenServer, still this
> would indicate a problem in my opinion.
>
>
>
> Thanks in advance for any assistance on this issue. Stepping back from the
> ledge now LOL
>
>
>
>
>
>
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-- 
[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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