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IP PHONE SERVICE CHOPPY

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mhodges

Technical User
Mar 6, 2008
142
US
OK how to explain this. This may be more difficult to figure out. We have a signaling server over at one of our other campuses that is hooked up to one of our main campuses. It is an IP enabled Option 61 hooked up thru a C6509. We also have a T1 that goes over to this campus. The problem is that on voice traffic every 3rd word on intermittent calls gets cut off. It can be inbound,outbound, 4 digit or 10. After awhile you can not continue and you have to hang up. Sometimes you have to wait like 10 min before it clears and you can call again. The back ups are done at night. I have thought of a few things like T1 traffic studies, sip channel study. I dont know where to start. Has anyone else had this happen? My boss changes the QOS but it is not remedied. It is not a specific time of day either that it happens. Any ideas?
 
I spotted this basic test in a post here a while back:

[tt]
This is by no means a fool proof method of qualifying a network for VoIP. However, this test will easily prove a network is NOT capable of supporting VoIP.

First all you need is a computer on the network running a MS Windows operating system.

Simple VOIP Network Test using a Ping Command

Instructions

* 1. Select Start > Run.
* 2. Type cmd, click OK.
* 3. At the prompt, type:
ping –t –l160 –w250 192.168.0.100
(VoIP Processor/Blade IP address, such as 192.168.0.100) and press Enter.

This starts the ping program and sends packets to the VoIP device address (or other test device).

–l160 makes the packets the same size as a 20msec. voice packet.

–w250 says if there is no response in 250 msec., the packet is dropped.

Any real-time application such as voice require no more than 250msec response time (I actually prefer no more than 200msec)

Periodically you can press CTRL + BREAK keys and see the packet loss statistics. Packet loss ratios of 1% or less can be considered acceptable.

This tool provides quantitative results that can be used when trying to engineer the network.
[/tt]

~~~
[small]GHTROUT.com | CS1000 Programming and Feature Guides | Tek-Tips FAQs | Recent Replies[/small]
 
OK i put in: ping (space) -t (space) -1160 (space) -w250(space) then the IP number and it says -1160 is a bad option. Eh....do I need all these spaces? maybe some commas? This pinging is new for me. LMK, Thanks,
 
id did L160 & it worked. My boss said it was and L, not a 1. I did not show any packet loss NOW.
 
Ryan, thanks for that. I also noticed it didn't work!

So, doing like this appears to work...

ping 192.168.1.30 -t -w250 -l160

Or whatever IP address.

mhodges, you would have to perform this "from" the site having trouble. In other words, you need to be there.

~~~
[small]GHTROUT.com | CS1000 Programming and Feature Guides | Tek-Tips FAQs | Recent Replies[/small]
 
actually that is wrong too . .

ping 192.168.1.30 -t -w 250

is right as you have to do the -w then a space then the timeout. I don't know where the -1160 comes into play. Needs a parameter I have a feeling. Here is the help from ping:

Usage: ping [-t] [-a] [-n count] [-l size] [-f] [-i TTL] [-v TOS]
[-r count] [-s count] [[-j host-list] | [-k host-list]]
[-w timeout] [-R] [-S srcaddr] [-4] [-6] target_name

Options:
-t Ping the specified host until stopped.
To see statistics and continue - type Control-Break;
To stop - type Control-C.
-a Resolve addresses to hostnames.
-n count Number of echo requests to send.
-l size Send buffer size.
-f Set Don't Fragment flag in packet (IPv4-only).
-i TTL Time To Live.
-v TOS Type Of Service (IPv4-only).
-r count Record route for count hops (IPv4-only).
-s count Timestamp for count hops (IPv4-only).
-j host-list Loose source route along host-list (IPv4-only).
-k host-list Strict source route along host-list (IPv4-only).
-w timeout Timeout in milliseconds to wait for each reply.
-R Use routing header to test reverse route also (IPv6-only).
-S srcaddr Source address to use.
-4 Force using IPv4.
-6 Force using IPv6.

 
OK i will go to the location after lunch when traffic picks up. I got it to work at the location where I am, but that is not where we are having the problem. I find it much easier troubleshooting voice than data packets. Ish....
 
GH, yeah the site was timing out so I just hit refresh after like an hour, and you had already posted the L . . . I'm an idiot!
 
ok either or they both seem to work. i am seeing 4 - 6 all the way up to 24ms, then I click control c and it says no packet loss. how long do i let it run for? do i make tests calls at the same time? what am i testing?
 
Well if you want to get creative you export it to a text file and run it over a 48-72 hour period, then see if there was any problems in that text file. If you want to go this route I can do all the commands you would need for a batch file just let me know.
 
my boss said he know how to do the text file to save. I called a few people. The ms went up a little but not much, then i press control c and it says lost 1 pack. no % lost. Eh....is there any way to see if i can max it out?
 
ok i have the media windows player going, goto all kinds of internet open, using the phone, both lines, and thing else I can try? LMK, TSM,
 
If your boss is the data wizard, have them map a few drives between PCs and start uploading and download huge blocks of files to saturate the network.

You don't need to do the ping test from one of those PCs though - you just need to be on the same part of the network.




~~~
[small]GHTROUT.com | CS1000 Programming and Feature Guides | Tek-Tips FAQs | Recent Replies[/small]
 
Several users said that it starts to happen after they have been on the phone for awhile like 2 mins. Somethough says it happens upon the initial call. It sounds like being in the bottom of a barrel. YIKES. I went over there and pinged like you said and never got it to go over 130ms. My boss can do the text file test if he wants I guess. Is there anything else you can think of that I can test. Let me know. Thanks,
 
Not that this is a T1 issue, but are your loops setup for 3VCE instead of just VOD or VCE?? I have heard of issues on inbound/outbound calls if it is not 3VCE.

Just something to try - you can always change it back.

PRI* TRK LOOP 105 - ENBL
FFMT/LCMT/YALMT: ESF/B8Z/FDL
SERVICE RESTORE: YES
YEL ALM PROCESS: YES
ALARM STATUS : NO ALARM
CH 01 - IDLE TIE 3VCE * CH 02 - IDLE TIE 3VCE *
CH 03 - IDLE TIE 3VCE * CH 04 - IDLE TIE 3VCE *
CH 05 - IDLE TIE 3VCE * CH 06 - IDLE TIE 3VCE *
CH 07 - IDLE TIE 3VCE * CH 08 - IDLE TIE 3VCE *
CH 09 - IDLE TIE 3VCE * CH 10 - IDLE TIE 3VCE *
CH 11 - IDLE TIE 3VCE * CH 12 - IDLE TIE 3VCE *
CH 13 - IDLE TIE 3VCE * CH 14 - IDLE TIE 3VCE *
CH 15 - IDLE TIE 3VCE * CH 16 - IDLE TIE 3VCE *
CH 17 - IDLE TIE 3VCE * CH 18 - IDLE TIE 3VCE *
CH 19 - IDLE TIE 3VCE * CH 20 - IDLE TIE 3VCE *
CH 21 - IDLE TIE 3VCE * CH 22 - IDLE TIE 3VCE *
CH 23 - IDLE TIE 3VCE * CH 24 - IDLE TIE 3VCE *
 
My boss said it was set to 3VCE. Also the location just called and said that it happened and my boss looked and he said only 1/2 meg was being used so he is thinking it is not the bandwidth. Eh....this is so frustrating. It never happens at any of our other offices. Could it be the signaling server? Eh....
 
Is there anyway to find out what other applications are running? I think some other applicatation that starts running is causing this. Eh.....
 
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