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<mailto:tjrlist at live.com> To: asterisk-users at lists.digium.com<mailto: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 -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/ff2ae758/attachment.html>
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. PeerCall IDDurationRecv: PackLost( %)JitterSend: PackLost(%)Jitterx.x.x.x5531341d06b00:07:4200000231230000063836(73.41%)0.000000000231020000000000(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 -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/f83d1bb6/attachment.html>
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 > > > > > > > -- _____________________________________________________________________ > -- 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 -------------- next part -------------- An HTML attachment was scrubbed... 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