On Tue, Jan 20, 2015 at 8:25 PM, tirveni yadav <[email protected]> wrote:
> > > On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog < > [email protected]> 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. <[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 >> >> > > 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.
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