Interesting Post. Asterix is great to terminate the VOIP calls, but
eventually it needs to be interfaced with a Gateway provider or the public
PBX network. What do you plan to use for that? You are not a CLEC are you?
RapidDSL & Wireless, Inc
IntAirNet- Fixed Wireless Broadband
----- Original Message -----
From: "Matt Larsen - Lists" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <email@example.com>;
<[EMAIL PROTECTED]>; "WISP-Related Topics" <[EMAIL PROTECTED]>
Sent: Tuesday, September 27, 2005 4:41 AM
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
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
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
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
5) AMP has a few bugs, namely with the provisioning of inbound numbers.
Everything looks right on the web page, but sometimes the configs work and
sometimes they do not. I have a trouble ticket in with them on my support
contract so that hopefully I can get to the bottom of the problem. Other
than this, AMP is an excellent front-end for asterisk and I am pretty
confident that I could put 100 or so users on this system and manage them
without a lot of problems. After that point it will be time for a real
server that is integrated into my billing and accounting systems, but for
experimenting and doing proof of case, this one will work just fine.
Conclusions so far:
- Asterisk is pretty decent for testing out voip and doing small scale
implementations. I'm pretty sure that it will scale a lot larger and do
more, but will also get more complex to manage
- Teliax is great for ITSP services and is very asterisk friendly.
- AMP is nice, but has a few bugs. The 1.10.008 version may be a bit
more stable than the latest version (1.10.009) but lacks a few important
features if you are intending to do some beta testing to customers or
- G729 codec is far superior to G711 over 802.11b wireless. GSM is also
good, but is not supported by many ATA adapters. ILBC is supposedly the
best one for wireless networks, but I have not had a chance to test it out
- Sipura adapters are inexpensive and have lots of features, but don't
have GSM or ILBC codecs. - Even under wireless conditions that are
pretty hostile (heavily loaded access point, 2.4Ghz cordless phone,
wireless connection inside the house) the voip quality was very
acceptable. - Even without QOS, voip was still usable. I would really
like to figure out how to optimize my StarOS AP units for VOIP, but
haven't found a good template to use.
I hope that this information is useful to some of you out there.
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