RE: [WISPA] Some VOIP Experimenting

2005-09-27 Thread Paul Hendry
Hi Matt,

We are using Asterisk too and it does seem to be very feature rich.
When you where testing did you notice a difference between the voice quality
of GSM and G729? Also, is your Asterisk server peering with your provider
using IAX2 and G711?
To optimize your StarOS AP I guess your gonna need to wait for V3 or
upgrade to StarVX ;)

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Behalf Of Matt Larsen - Lists
Sent: 27 September 2005 09:41
To: [EMAIL PROTECTED]; wireless@wispa.org; [EMAIL PROTECTED];
WISP-Related Topics
Subject: [WISPA] Some VOIP Experimenting

Hello all,

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 
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.   
http://www.teliax.com/
-  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 
reselling.
-  

RE: [WISPA] Some VOIP Experimenting

2005-09-27 Thread Brian Webster
Matt,
I found an easy way to set up an asterisk box with AMP called [EMAIL 
PROTECTED]
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.



Thank You,
Brian Webster
www.wirelessmapping.com http://www.wirelessmapping.com
Free World Dialup #481416


-Original Message-
From: Matt Larsen - Lists [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 27, 2005 4:41 AM
To: [EMAIL PROTECTED]; wireless@wispa.org; [EMAIL PROTECTED];
WISP-Related Topics
Subject: [WISPA] Some VOIP Experimenting


Hello all,

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
numbers.