Matt,

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?

Tom DeReggi
RapidDSL & Wireless, Inc
IntAirNet- Fixed Wireless Broadband


----- Original Message ----- From: "Matt Larsen - Lists" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; <wireless@wispa.org>; <[EMAIL PROTECTED]>; "WISP-Related Topics" <[EMAIL PROTECTED]>
Sent: Tuesday, September 27, 2005 4:41 AM
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. - 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 yet. - 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.

Matt Larsen
[EMAIL PROTECTED]


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