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[asterisk-users] sip show channelstats reliable?


 
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tjrlist at live.com
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PostPosted: Mon Jan 19, 2015 1:17 pm    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

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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tjrlist at live.com
Guest





PostPosted: Mon Jan 19, 2015 1:45 pm    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

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 Smile


From: tjrlist@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
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EWieling at nyigc.com
Guest





PostPosted: Mon Jan 19, 2015 1:56 pm    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

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@lists.digium.com [mailto:asterisk-users-bounces@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 Smile



From: tjrlist@live.com (tjrlist@live.com)
To: asterisk-users@lists.digium.com (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
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tjrlist at live.com
Guest





PostPosted: Mon Jan 19, 2015 2:01 pm    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

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@nyigc.com
To: tjrlist@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抳e 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抣l try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn抰 work.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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 Smile



From: tjrlist@live.com (tjrlist@live.com)
To: asterisk-users@lists.digium.com (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
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sgriepentrog at digium...
Guest





PostPosted: Mon Jan 19, 2015 2:13 pm    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

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@live.com (tjrlist@live.com)> wrote:
Quote:
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@nyigc.com (EWieling@nyigc.com)
To: tjrlist@live.com (tjrlist@live.com); asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats reliable?


I鈥檝e seen something similar with Adtran SIP gateways.聽聽聽 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.聽 聽聽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.聽聽聽聽 At some point I鈥檒l try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn鈥檛 work.

From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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
(聽聽聽聽 %)
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 Smile



From: tjrlist@live.com (tjrlist@live.com)
To: asterisk-users@lists.digium.com (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:
聽 聽 聽 聽 聽 聽 聽 聽http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
聽 聽http://lists.digium.com/mailman/listinfo/asterisk-users




--
Scott Griepentrog
Digium, Inc 路 Software Developer
445 Jan Davis Drive NW 路 Huntsville, AL 35806 路 US
direct/fax: +1 256 428 6239 路 mobile: +1 256 580 6090
Check us out at: http://digium.comhttp://asterisk.org
Back to top
yadav.tirveni at gmail...
Guest





PostPosted: Tue Jan 20, 2015 9:56 am    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog@digium.com (sgriepentrog@digium.com)> wrote:
Quote:
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@live.com (tjrlist@live.com)> wrote:
Quote:
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@nyigc.com (EWieling@nyigc.com)
To: tjrlist@live.com (tjrlist@live.com); asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats reliable?


I鈥檝e seen something similar with Adtran SIP gateways.聽聽聽 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.聽 聽聽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.聽聽聽聽 At some point I鈥檒l try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn鈥檛 work.

From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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
(聽聽聽聽 %)
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 Smile



From: tjrlist@live.com (tjrlist@live.com)
To: asterisk-users@lists.digium.com (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:
聽 聽 聽 聽 聽 聽 聽 聽http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
聽 聽http://lists.digium.com/mailman/listinfo/asterisk-users






--
Scott Griepentrog
Digium, Inc 路 Software Developer
445 Jan Davis Drive NW 路 Huntsville, AL 35806 路 US
direct/fax: +1 256 428 6239 路 mobile: +1 256 580 6090
Check us out at: http://digium.comhttp://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聽 date; asterisk -rnx 'sip show channelstats';聽 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聽 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聽 sip_client_ip -> asterisk_server_ip聽 RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=314, Time=50240
1036396 2000-01-03 22:08:11.647404聽 sip_client_ip -> asterisk_server_ip聽 RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=383, Time=61280
1036401 2000-01-03 22:08:11.647560聽 sip_client_ip -> asterisk_server_ip聽 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

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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yadav.tirveni at gmail...
Guest





PostPosted: Sat Jan 24, 2015 2:33 am    Post subject: [asterisk-users] sip show channelstats reliable? Reply with quote

On Tue, Jan 20, 2015 at 8:25 PM, tirveni yadav <yadav.tirveni@gmail.com (yadav.tirveni@gmail.com)> wrote:
Quote:


On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog@digium.com (sgriepentrog@digium.com)> wrote:
Quote:
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@live.com (tjrlist@live.com)> wrote:
Quote:
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@nyigc.com (EWieling@nyigc.com)
To: tjrlist@live.com (tjrlist@live.com); asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats reliable?


I鈥檝e seen something similar with Adtran SIP gateways.聽聽聽 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.聽 聽聽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.聽聽聽聽 At some point I鈥檒l try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn鈥檛 work.

From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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
(聽聽聽聽 %)
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 Smile



From: tjrlist@live.com (tjrlist@live.com)
To: asterisk-users@lists.digium.com (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







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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聽 date; asterisk -rnx 'sip show channelstats';聽 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聽 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聽 sip_client_ip -> asterisk_server_ip聽 RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=314, Time=50240
1036396 2000-01-03 22:08:11.647404聽 sip_client_ip -> asterisk_server_ip聽 RTP PT=ITU-T G.711 PCMA, SSRC=0x488EDB49, Seq=383, Time=61280
1036401 2000-01-03 22:08:11.647560聽 sip_client_ip -> asterisk_server_ip聽 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?


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