[asterisk-users] sip show channelstats reliable?

[tirveni yadav]

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
>>>
>>>
>>>
>>>
>>>
>>>
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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.
>
>
>
>


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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