On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepent...@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. <tjrl...@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: ewiel...@nyigc.com >> To: tjrl...@live.com; asterisk-users@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-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@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: tjrl...@live.com >> To: asterisk-users@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. -- Regards, Tirveni Yadav www.udyansh.com <http://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.
-- _____________________________________________________________________ -- 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