I found an easy way to set up an asterisk box with AMP called [EMAIL
http://asteriskathome.sourceforge.net/, they have good help pages. You burn
an ISO disk and then let it do it's thing on boot. It will wipe the hard
drive clean and do a fresh install but it works great. I ran it on a similar
machine to yours. I'll have to play with the codec's some more to see about
performance issues. I just picked up a USB phone and have started playing
with it in addition to my sipura box. On a laptop it's nice not needing the
headset. When I set mine up I put it in the DMZ of my router, when I was on
the road helping Mac I was able to get the IAX soft phone to connect as an
extension. This allowed me to pull dial tone from home wherever I could
connect, nice tool. I have a VOIP account from Broadvoice and was able to
log in and switch over easily to my asterisk box from the SIPURA I was
using. For those who don't want to set up a paid account with a VOIP
provider they can set up a Free World Dial Up account
http://www.freeworlddialup.com/. With this you can call peer to peer,
outgoing 800 numbers, and PSTN callers can call you if they have the list of
access numbers attached all for free. I encourage everyone to at least play
with this and become familiar with the technology. VOIP has been a big part
of the hurricane recovery efforts for WISP's.
Thanks for the information Matt, it will be a big help to all.
Free World Dialup #481416
From: Matt Larsen - Lists [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 27, 2005 4:41 AM
To: [EMAIL PROTECTED]; firstname.lastname@example.org; [EMAIL PROTECTED];
Subject: [WISPA] Some VOIP Experimenting
I've been doing some experimenting with VOIP and asterisk over my
wireless network and came up with a few general observations that I
thought might be useful for those of you looking at voip and how it
impacts your network:
1) My test voip server is a P3 550 with 512MB of memory. I installed
Fedora Core 3 and AMP (Asterisk Management Portal - available from
http://amp.voxbox.ca). This provides a very straightforward web based
interface for configuring Asterisk, but is not without a few bugs - more
on that later.
2) I am using accounts from Nufone and Teliax to do my beta testing.
Nufone has been around longer and has a reputation for being very solid
technically, but does not have much for online assistance. Teliax has
excellent online resources, and also has local numbers in many places.
I was quite surprised to find out that they had local numbers in my
small town in Nebraska. Nufone works, but I think that Teliax may be
worth a little extra just to get the better support resources and access
to more local numbers.
3) I am using a Sipura SPA-2000 two line adapter at home to do my
testing. With some experimentation, I was able to get this adapter to
work through NAT.
4) My home connection is limited to 1024K down/512K upload and connects
to a StarOS access point. My home CPE is a WRAP board with a CM9 card
running in 802.11b mode. The StarOS AP has 200+ customers on it, is
located 8 miles away and has approximately 80 customers on the sector
that services my house. The VOIP server is two hops from my home, and
average latency to it is 10ms. There is no QOS on the network. The
telephone in the house is a Panasonic 2.4Ghz Frequency hopping phone,
and it sits next to a Ezy Net radio in client mode that connects to my
home access point. I use an IPCop firewall box. The IPCop box is an
older version that doesn't have the QOS shaping available.
4) Initial testing was with the G711 (aka ulaw) codec that is standard
on the Sipura and also a standard codec in Asterisk. This codec
used 80KB up , and about 80KB down when a two way conversation was
going. On this heavily loaded AP, this was a bit of a problem, but it
was usable. I was able to carry on a one hour conversation with only a
minimum of noticeable breakup one night, and the next day I had another
conversation that deteriorated rapidly. Downloading also seemed to
affect the connection quite a bit.
5) Second round of testing was with an iaxComm softphone. The
softphone connected with the GSM codec and used 15 to 25KB during
conversation. Audio quality seemed to be pretty good, but it was hard
to tell becuase I do not have a headset on my PC - I was dependent on
the built-in speakers and microphone.
6) Final round of testing was with the Sipura adapter after I was able
to get the G729 codec operational on the Asterisk box. By setting the
Sipura to only use G729, traffic was 24KB up and 24KB down during two
way conversation. Audio quality was not quite as good as with G711,
but there were fewer breakups and even with a large download going, it
was still usable.
5) AMP has a few bugs, namely with the provisioning of inbound