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VoIP 150 ms

Posted on 2013-01-20
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Last Modified: 2013-02-04
It says from the CCNP voice study that the max end-to-end transit time for voice is 150ms. How do I test that? The reason I ask is because my users are complaining of internittent lost of voice during their VoIP phone conversation. Thanks
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Question by:biggynet
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7 Comments
 
LVL 10

Expert Comment

by:joelsplace
ID: 38799829
Ping the remote phone system from the phone's location or vise-versa.  Ping -t will keep it going and show times in ms.
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Author Comment

by:biggynet
ID: 38799967
Is ping a reliable tool for testing latency?
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LVL 35

Assisted Solution

by:Ernie Beek
Ernie Beek earned 125 total points
ID: 38800637
Ping gives you a fair insight of the latency between two nodes.

You can also use pathping which is a combination of ping and tracert and provides you more info of the intermittent nodes (hops) as well.

pathping 213.201.128.21

Tracing route to ns1.novaxess.nl [213.201.128.21]
over a maximum of 30 hops:
  0  xxx.xxx.local [x.x.x.x]
  1  x.x.x.x
  2  nl-rt-dc2-ias-arg27.kpn.net [62.12.4.63]
  3  nl-asd-dc2-ice-ir01.kpn.net [139.156.222.65]
  4  te3-0-0.gr10.saams.nl.easynet.net [195.69.144.38]
  5  ge2-1-111.br11.saams.nl.easynet.net [87.86.71.209]
  6  ns1.novaxess.nl [213.201.128.21]

Computing statistics for 150 seconds...
            Source to Here   This Node/Link
Hop  RTT    Lost/Sent = Pct  Lost/Sent = Pct  Address
  0                                           xxx.xxx.local [x.x.x.x]
                                0/ 100 =  0%   |
  1    0ms     0/ 100 =  0%     0/ 100 =  0%  x.x.x.x
                                0/ 100 =  0%   |
  2    6ms     0/ 100 =  0%     0/ 100 =  0%  nl-rt-dc2-ias-arg27.kpn.net [62.12.4.63]
                                0/ 100 =  0%   |
  3   13ms     0/ 100 =  0%     0/ 100 =  0%  nl-asd-dc2-ice-ir01.kpn.net [139.156.222.65]
                                0/ 100 =  0%   |
  4   11ms    93/ 100 = 93%    93/ 100 = 93%  te3-0-0.gr10.saams.nl.easynet.net [195.69.144.38]
                                0/ 100 =  0%   |
  5  ---     100/ 100 =100%   100/ 100 =100%  ge2-1-111.br11.saams.nl.easynet.net [87.86.71.209]
                                0/ 100 =  0%   |
  6    9ms     0/ 100 =  0%     0/ 100 =  0%  ns1.novaxess.nl [213.201.128.21]

Trace complete.

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LVL 21

Accepted Solution

by:
robocat earned 250 total points
ID: 38800641
Ping is NOT a reliable tool for testing latency in a VOIP environment. Ping is often treated as low priority network traffic by WAN networks and the results are not in any way representative for VOIP traffic.

Most VOIP systems do collect extensive data themselves (latency, jitter, packet loss), so the best way to do this is get the information from within the ip phones themselves. How to do this depends on your specific brand.
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Author Comment

by:biggynet
ID: 38801446
I am fairly new to VoIP. But I have Avaya VoIP phones. So I can get the stats from within the phone or it is somewhere in the PBX.
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LVL 21

Assisted Solution

by:robocat
robocat earned 250 total points
ID: 38801520
Both the phones and the PBX should be able to deliver some stats.

Avaya has a software called "Avaya Prognosis VOIP Monitor" that facilitates accessing these stats.
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LVL 9

Assisted Solution

by:user_n
user_n earned 125 total points
ID: 38802118
The problem may be in jitter buffer configuration
http://en.wikipedia.org/wiki/Jitter
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