hello,
There are a couple of things you can do to test the route between you and your ua. First, go to the sip server and look at the registered users screen and then find the ip address of your user. Then go to msdos promt and type ping -t 11.22.33.44 where 11.22.33.44 is the ip address you got from the registered screen.
Let it run a while and check the time it shows which represents how long a packet takes to go to the ua and back.
If you see a time > 100ms then type traceroute 11.22.33.44 and watch all the routers between you and the ua to see where the bottleneck is. If the first hop is > 100ms then the problem is on the server side, if the end point is > 100ms then it's the ua's internet connection problem.
You can download pingplotter free trial pro to get a very good picture in graph form of what i described above.
You are looking for the pings to be somewhat consistant. If some are 30 and some are 100 and it changes around often, then you will have poor voice quality because even if the ua can adjust the jitter buffers, it can't adjust them very fast so fluctuating latency(jitter) can cause some of the conversation to break up.
All that info above only tells you if your connection is stable, but not if you have a bandwidth problem.
G711u and a conversations take about 80kbs. 64kbs for the data and about 16kbs for the packet header information that helps the data find its way to the ua. To be safe, always calculate 100kbs per conversation to give some room for error. We have found that any internet connection can only do voice with quality up to about 80-85% of it's maximum speed. So you need about 125kbs minimum upload and download speed for 1 conversation.
To check your bandwidth, you should have both the ua internet and the server internet run a bandwidth test.
You can do this by going to
www.speedtest.net. this will tell you if you have enough bandwidth to carry on a good conversation. Also on the the speedtest site is a button called ping test. Click that after you finish the bandwidth test and you can test for jitter and packet loss. A very exellent site for us voip people. If you would like to report back your findings we would be happy to help you interpret the results.
One last thing, if your ua supports ilbc, by all means use it if you are short on bandwidth. It greatly reduces the amount of bandwidth that a conversation uses and sounds so clear you most likely can't hear the difference between g711u and ilbc. The only reason more people don't use it is because it takes a little more processing power than g729. For us with Brekeke with the small number of simultaneous conversations you probably will have it's fine. For large service providers carrying many simultaneous conversations it may use too much processing power so they like g729 which is less cpu intensive.
Another diagnostic tool is wireshark. You can run wireshark(free) on your server and capture all the packets of a conversation with a click or two. Then when the conversation is over, you can click on tools and get a great report on the conversation that would include, latency, packet loss and jitter. With the proper tools, you can always see exactly what a sip call problem is and make sure you have high quality. It's not guesswork, it's just work.
