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Forest Park, IL

[Fixed] WISP latency causing poor voip call quality

Please pardon my technical inexperience up front. I am attempting to educate myself.

I had a WISP install a 5/1Mbps internet connection to a customer site in Las Vegas.

The connection was installed to support two IP phones supporting 4 concurrent calls.

VOIP provider is telling me that there is too much latency to support good call quality.

when I ping the phones from my PC, in Chicago, I get response somewhere between 80 and 90ms.

Is there any simple fix to improve the latency?



Queen Anne, MD
Run a test at and and post the results.


Forest Park, IL
Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

C:\Documents and Settings\Administrator>ping

Pinging with 32 bytes of data:

Reply from bytes=32 time=97ms TTL=47
Reply from bytes=32 time=89ms TTL=47
Reply from bytes=32 time=96ms TTL=47
Reply from bytes=32 time=89ms TTL=47

Ping statistics for
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 89ms, Maximum = 97ms, Average = 92ms

I am not at the location so I cannot do a speed test


·Montrose Mutual

1 edit
most WISP have 80-90ping which really isn't that much of a delay and should work fine with VoIP.

My guess is you need more upload. 1mb is really 120~kb/s split 4 ways is only 30kb/s which may not be enough for your VoIP quality setting.

*that's assuming nothing else is running and your getting full speed.

/edit I could be wrong I don't use VoIP :/ » ··· 127.html

"Depending on the protocol used, each concurrent call requires a bandwidth of between 30Kbps to 80Kbps."

" A low-end 2Mbps download and 512Kbps upload broadband connection should be enough for two or three concurrent calls"


reply to palock34
4x 64kbit voip streams should easily fit into the 1mbit upload.

I would ask if the WISP is using TDMA. If not, then voip probably wouldnt work over the connection when the tower is under a small amount of load and I would suggest giving up now.

If they do use TDMA then you will want to look into your jitter.
Linksys voip phones have an adjustable jitter buffer - i just set mine to "extremely high" and it can move "up only", rather than "up and down" and it works fine.

90ms isnt really a problem. 500ms+ will be noticable as a small delay when talking - like watching a live news reporter being interviewed by the studio host via satelite link. But the stream should still be constant and work fine if the jitter buffer is working.


Siren, WI
reply to Org
said by Org:

1mb is really 120~kb/s split 4 ways is only 30kb/s which may not be enough for your VoIP quality setting.

You are mixing up kilobits (kb) with kilobytes (kB). 1 megabit is 1024 kilobits, or 128 kilobytes. A VOIP call at most should be 64 kilobits or 8 kilobytes. 1 meg should theoretically support 16 simultaneous 64kb phone calls, but in reality wouldn't leave any headroom. Either way, a few simultaneous calls should be fine.