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3Com NBX 100 over 2 T1 lines

I am having a problem with a remote site. I have 35 users with 3com VoIP at the location. So they have 35 computers running over the network and 35 phones over the network. I currently have 2 dedicated T1's going to that site. They are having problems with voice cutting out while you are on the phone. The phone conversation is choppy.
1 Solution
Hi gbauer17,
This is likely to be a problem with packet prioritisation, however it could be down to sheer bandwidth.

Use the bandwidth calculator at http://www.voip-calculator.com/calculator/lipb/ to help you work out how much bandwidth you need between the sites.
If you had 35 calls happening at once, using G.711 (uncompressed data), then you'd be using 2880 kbps.  I'd imagine that you're using a compressed codec though!

VoIP gets very fussy about round-trip latency, whereas most data can cope with a few milliseconds delay.

Do you have routers at each end that are capable of traffic shaping/prioritisation?  You need to configure these to prioritise the VoIP traffic

It's worth doing some end-to-end ping tests to check what your latency is and make sure that you have no packet loss.
I suggest taking a look at http://www.voip-info.org/tiki-index.php?page=QoS for more information.

Does that help?

gbauer17Author Commented:
I know that 35 calls at once can produce that problem. But I am pretty sure that not all people are on the phone at once. We are not using any compression though. Any suggestions on what we could use.
This is a good summary of codecs :-

I would suggest the Speex codec if supported as you can vary the bandwidth it uses. ILBC and GSM are reasonable. G729 is excelent for the bandwidth it uses but requires a commercial license which might be included in your product.

Note that if you make data calls over the voice network (faxes for example) they you need to stick with G711 for these as other codecs compress the audio too much and stop them working.

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