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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On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog at digium.com> wrote:> 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 > >You can find out the data loss outside of Asterisk by using tcpdump and tshark(wireshark) 1. Capture output of Asterisk SIP channels in a log file ax_log_yyyymmdd $while :; do date; asterisk -rnx 'sip show channelstats'; sleep 5 ; done>> ax_log_yyyymmdd2. Capture tcpdump traffic on the asterisk server: $tcpdump -nq -s 0 -i eth0 -G3600 -w eth_sip_traffic-%F-%H-%M-%S.pcap port 5060 or port 5061 [this saves the all the ethernet traffic of ports 5060 & 5061 in the pcap file for every hour(-G 3600) ] 3. Once you can see the data loss in the ax_log_yyyymmdd, check for the same time in the eth_sip_traffic.pcap Analyze the eth_sip_traffic.pcap $tshark -t ad -r eth_sip_traffic.pcap |grep sip_client_ip | less [ -t ad: is for time format, -r :is for input file] 1034847 2000-01-03 22:08:10.239661 sip_client_ip -> asterisk_server_ip RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=314, Time=50240 1036396 2000-01-03 22:08:11.647404 sip_client_ip -> asterisk_server_ip RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=383, Time=61280 1036401 2000-01-03 22:08:11.647560 sip_client_ip -> asterisk_server_ip RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=384, Time=61440 You can find the if the packets loss is happening, with the missing sequence numbers. PS: I think any loss greater than 3%, will deteriorate the call quality. -- Regards, Tirveni Yadav www.udyansh.com <http://www.udyansh.org> What is this Universe ? From what it arises ? Into what does it go? 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On Tue, Jan 20, 2015 at 8:25 PM, tirveni yadav <yadav.tirveni at gmail.com> wrote:> > > On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog < > sgriepentrog at digium.com> wrote: > >> 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 >> >> > > You can find out the data loss outside of Asterisk by using tcpdump and > tshark(wireshark) > > 1. Capture output of Asterisk SIP channels in a log file ax_log_yyyymmdd > > $while :; do date; asterisk -rnx 'sip show channelstats'; sleep 5 ; done > >> ax_log_yyyymmdd > > 2. Capture tcpdump traffic on the asterisk server: > > $tcpdump -nq -s 0 -i eth0 -G3600 -w eth_sip_traffic-%F-%H-%M-%S.pcap port > 5060 or port 5061 > [this saves the all the ethernet traffic of ports 5060 & 5061 in the pcap > file for every hour(-G 3600) ] > > 3. Once you can see the data loss in the ax_log_yyyymmdd, check for the > same time in the eth_sip_traffic.pcap > > Analyze the eth_sip_traffic.pcap > > $tshark -t ad -r eth_sip_traffic.pcap |grep sip_client_ip | less > [ -t ad: is for time format, -r :is for input file] > > 1034847 2000-01-03 22:08:10.239661 sip_client_ip -> asterisk_server_ip > RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=314, Time=50240 > 1036396 2000-01-03 22:08:11.647404 sip_client_ip -> asterisk_server_ip > RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=383, Time=61280 > 1036401 2000-01-03 22:08:11.647560 sip_client_ip -> asterisk_server_ip > RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=384, Time=61440 > > You can find the if the packets loss is happening, with the missing > sequence numbers. > > PS: I think any loss greater than 3%, will deteriorate the call quality. > > > >Is it possible that this kind of packet loss in sip channels can cause High load on the server? -- Regards, Tirveni Yadav www.udyansh.org What is this Universe ? From what it arises ? Into what does it go? In freedom it arises, In freedom it rests and into freedom it melts away. Upanishads. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150124/c9678a81/attachment.html>