[asterisk-users] sip show channelstats reliable?

tirveni yadav yadav.tirveni at gmail.com
Tue Jan 20 08:55:07 CST 2015


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
>>
>>
>>
>>
>>
>>
>> -- _____________________________________________________________________
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>>
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>
>
>
> --
> [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.
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