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. <[email protected]> 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: [email protected] > To: [email protected]; [email protected] > 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:* [email protected] [mailto: > [email protected]] *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: [email protected] > To: [email protected] > 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 > > > > > > > -- _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New > to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE > or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- [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
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
