My VOIP in my business is failing and I don't know why

Hello I have a question regarding my VOIP
it is very strange and I cannot for the life of me figure this out.

To begin my VOIP was working perfectly fine. Now it is about 2 weeks that it is acting up. Here is the problem

At times my call will completely disconnect
the most common problem would so far is the person on the other end can hear me but I cannot hear them

it does happen that i hear them but they cannot hear me.

Here are some things I have tried to change since I have had this problem.

I changed my switch I now have a cisco 24port switch
I have changed my VOIP provider
I have a program called softtalk on my pcs that i have tried
I have formatted all the pcs (only 10) and put them back to stock.

Nothing seems to work what I cannot understand is my old system that was working fine (a northstar system) now has the same symptoms as my new system.

I changed all the wiring in my building and I still have the same problem.

My ISP sees 0 packet loss on my line.

I am completely stumped on this iI have no idea what my problem can be can someone please give me a little insight on this?

thanks a bunch!

Welcome to Voipuser forums.

It would help if you provided a little more information about your installation. Are you using a business line, what speed, what kind of network, pbx, and routers etc.. are all the computers on the same network or subnet? Who is the Voip provider? Is it just the one computer or several that run voip? If you make sure that only one computer is running voip then does it work OK? It sounds like you have a bumper problem with ports.

Ok so
I have 2 completely different systems

1 system I am using is Softphone and my provider for that is panterranetworks

my other system uses a Linksys spa2102 and the provider is teliphone.ca

both are giving the same problem and it just started happening out of the blue there were no changes done on my network that would be considered "out of the ordniary"

No the drops happen completely randomly my users are only allowed to go on 2 sites

1 is my website and the other is my management site.


My Inet line is 8mbps
all the computers have a manually assigned IP

Your SPA2102 has two voip lines and buit in router set to ports 5060 and 5061 UDP + needing some RTP ports UDP. The computers using softphone will need to use different ports. Is it plugged into the switch, the modem or a router? Can you explain what it is that you home to do with you network. It seems to me that you need some kind of pbx either hosted, hardware or and Asterisk box.

Ok my SPA2102 and softphone are 2 different systems

what is happening is my system with the SPA2102 is brand new I just got it a few days ago so it was never working at the same time as my softphone system.

However now I had them working together at the same time.

What I do is call people in the US my company is a company that deals with insurances so we dial out to the states everyday.

Now my linksys2102 is connected directly into my switch which is a
Cisco Business series 24 port SR224



To give you an idea of how my config is

I have my ISP Modem which gives me my static IP
connected to
A normal Linksys router to have internet
connect to
My cisco Switch


this is how the computers in my network receive internet. I dont know if that will help you out but I am trying to give you as much information as possible.
It is about 2 hours now that I disabled QoS on 1 of the pcs and I have not dropped a call but I really don't think that will be the explaination to my issues I just think it is a fluke that I have not dropped any calls.

Hi I may be missing something here, But how many calls are being made at the same time?

Also run a speed test such as http://myvoipspeed.visualware.com/index.html picking a location close to your normal destinations. then post the results, (Click Advanced, then show text )

you should get something like

VoIP test statistics
--------------------
Jitter: you --> server: 1.5 ms
Jitter: server --> you: 21.6 ms
Packet loss: you --> server: 0.0 %
Packet loss: server --> you: 0.0 %
Packet discards: 0.0 %
Packets out of order: 0.0 %
Estimated MOS score: 3.8

Ian

The most calls that can be made at exactly the same time is 14 about a month ago but now the most I do is 9

On a given day I can make a total of about 200 calls

VoIP test statistics
--------------------
Jitter: you --> server: 1.4 ms
Jitter: server --> you: 4.0 ms
Packet loss: you --> server: 0.0 %
Packet loss: server --> you: 0.0 %
Packet discards: 0.0 %
Packets out of order: 0.0 %
Estimated MOS score: 4.0

^^^^^^^This was your test^^^^^

while you were probably writing this I looked into Jitter and I did another test (not the one you sent me these were my results)

Test Detail Result
Download Speed: 11,775,560 bps
Upload Speed: 691,784 bps
QOS: 81%
RTT 87 ms
MaxPause: 72 ms
Jitter: 3813
Packet Loss: 0%
Download Jitter: 100%
Download Packet Loss: 1.7466 ms
Discards: 0%
Order: off
Test Result #: Off

I did another test from another site
this is my result

Code:

ownload speed: 16726624 bps Upload speed: 764904 bps Quality of service: 46 % Maximum TCP delay: 18 ms Average download pause: 1 ms Minimum round trip time to server: 12 ms Average round trip time to server: 19 ms   VoIP ---- Jitter: you --> server: 15.2 ms Jitter: server --> you: 15.1 ms Packet loss: you --> server: 0.0 % Packet loss: server --> you: 0.0 % Packet discards: 0.0 % Packets out of order: 0.0 % Number of VoIP lines supported: 5 Estimated MOS score: 3.7   Speed test statistics --------------------- Download speed: 16726624 bps Upload speed: 764904 bps Quality of service: 46 % Download test type: socket Upload test type: socket Maximum TCP delay: 18 ms Average download pause: 1 ms Minimum round trip time to server: 12 ms Average round trip time to server: 19 ms Estimated download bandwidth: 19200000bps Route concurrency: 1.1478707 Download TCP forced idle: 0 % Maximum route speed: 43690000bps   VoIP test statistics -------------------- Jitter: you --> server: 15.2 ms Jitter: server --> you: 15.1 ms Packet loss: you --> server: 0.0 % Packet loss: server --> you: 0.0 % Packet discards: 0.0 % Packets out of order: 0.0 % Estimated MOS score: 3.7   Streaming video statistics -------------------------- Audio stream jitter: 54.2 ms Video stream jitter: 54.3 ms Audio stream packet loss: 0 % Video stream packet loss: 0 % Audio stream packet discards: 0 % Video stream packet discards: 0 %   Capacity test statistics ------------------------ Download capacity: 5493232 bps Download packets per second: 4291 Upload capacity: 600960 bps Upload packets per second: 469 Quality of service: 95 % Packet size: 160 Bytes

from site voiptest.nuvio.com

If you might make 9 simultaneous calls then your upload speed is too slow (only slightly faster than mine 523 kbits/sec). For voip you need high speed (in your case perhaps 3000 kb/s depending on codec) both ways) to be assured of business reliable service. You could try a lower bandwidth codec but that might lower the call quality.

y would it just stop working though I was making 14 calls about a month ago
now i went down to abou 7 or 8 and it is giving me this problem?

also

how do i know what codec i am using?

anthonyaudi :

how do i know what codec i am using?

You can see that in the settings for the ATA 2102 and probably in the soft phone. The usual one the voip providers use is G 711 that give high quality. G 729 uses less bandwidth but is not always available.
http://www.voip-info.org/wiki-Codecs

I called my ISP
This is what they told me

They can see packet loss however they cannot explain it. They see packet loss after they reach the modem with their tests but they cannot explain why I am having this packet loss

I had them come and change the modem that gives me my static IP today
I will let you guys know if it is still going down.

As you have wrote in your initial post, you have changed everything except your ISP.

Change your ISP and your problems will disappear

This story has gone a bit cold not but Istvan might be right. I looks like you are using Bell Canada as ISP. They are well known to restrict bandwidth, especially on peer-peer activity.
http://www.wolfmanzbytes.com/news/viewn ... EZiyHaAuzD
(there are many other references). Rogers is also reported to do this. With the low upload speed that you have any throttling in that area will kill voip or at least make it very unreliable.