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VSAT
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STRANGE VOIP PERFORMANCE

Sat Oct 21, 2006 10:07 pm

HI all
We use an outdoor routerboard 532,Atheros 65mw card and 1W amp with 15DBi omni antenna
Network is set to B mode only on a frequency used only by us locally (however pleanty of other wifi traffic on other frequencies)
We currently have around 30 clients using the system all sharing 4Mbit connection and limited to 128k/256k or less
I have recently set up an asterisk box to be able to give voip service to our clients.
The problem is that I am getting some very strange behaviour from our network. (on the wired network works perfectly)
Before giving the service to our clients i want to make sure it works, but during testing, when I call from one extension to another (in the same network) or even if i dial the voicemail on the server i get mixed quality results.
Sometimes it is perfect, other times the voice is all broken up. There is no real pattern to it.
I have tried to prioritise VOIP traffic and limit P2P to the limit but what really worries me is that we are not even trying to call outside of the network.
All our ping times are good 5 to 7 ms on average and my personal home connection from where i do some of the testing is -54 with 35snr.
After reading many of the posts I realise that the use of an amp is not recommended (especially with atheros cards) but without it we dont have a very good range. Even with the amp our clients are all less than 1 km away.
Could the amp be distorting the VOIP traffic? If so why at times (few hours) it is perfect?
Any suggestion is welcome and much appreciated

Thanks
David
 
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Sun Oct 22, 2006 7:14 am

Hi
Take off the amp and put up more high sites.
 
DirectWireless
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Sun Oct 22, 2006 9:12 am

Not to mention CPE to CPE performance can be limited due to the fact that it's going from the CPE, to the AP, and to another CPE. So you run into packet collisions constantly. The reason for this is your SIP phones are directing the calls to each other directly, bypassing your QOS / traffic shaping because the AP is allowing them to forward packets directly.

You may be better off to turn "canreinvite=no" in your SIP.CONF clients to force them to only talk to the Asterisk server, then your QOS should stay working. But I recommend that you do take the amp off and switch to a SR2 card (400mW). You may also benefit from using 2ghz-10mhz which actually has more throughput and less interference than 802.11b, but requires ALL Mikrotik clients. We just recently took out a whole bunch of amps and switched to SR2's, and found we had much better signals, faster clients and our overall network was online more.

Lastly, for 30 clients of VOIP - you really should sectorize with 3 radios. We used to use 15db omnis, and now use 16db sector panels, and find that the sectors have way better coverage area over the omni. Also, your SIP to SIP calls will benefit if they are on different sectors - less collisions.
 
VSAT
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Sun Oct 22, 2006 8:05 pm

Hi
Thanks for the advice.
All SIP extensions are set to No "canreinvite".
I think our main problem is the amp and the number of registered users on the same antenna.
Am i right in thinking that I can set just one rule in Mangle to monitor and prioritise my Voip traffic just with my Asterisk server ip address?
Or would I have to set rules for each of the UDP ports used?

I think our best solution for the time being is to set up all clients that also use our voip service on a separate antenna thus splitting the load a little.

Can anyone that also gives voip service add any further advice?

Thanks

David
 
DirectWireless
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Sun Oct 22, 2006 10:24 pm

2 mangle rules, one for packets going TO your Asterisk server, one for packets coming FROM your Asterisk server. No need to worry about VOIP. Just make sure you have set up one main queue for each interface, and 2 (or more) queues - one for VOIP, one for everything else. Set your everything else limit way below 5.5Mbps, maybe 1.5Mbps, and use bursting to enable faster speeds. Set no limit on VOIP, and a max limit on your interface at around 4-5Mbps.

I'm going to VOIP and every place I can I'm moving to either 2ghz-10mhz or 5ghz both with NStreme & Polling. I'd say NStreme could help VOIP quite a bit because of the packet aggregation, but I'm using a standard 802.11b client (CB3) on this sector using a VOIP ATA and it's okay (not great but okay). I am deploying commerical / business VOIP in the next few weeks but don't know for sure if I'm going to continue using MT to do it after the experiences (and lack of other people's experiences). I need to support about 25-50 calls per AP and I think I will need a bigger system like Wimax to really do it well.
 
bushy
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Sun Oct 22, 2006 11:11 pm

You could have something wrong with your antenna/coax/radiocard , otherwise you wouldn't need the amp. ..Maybe there is non-wifi interference you are not picking up. ( may explain why no-one else is using that channel ).
 
phendry
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Wed Nov 08, 2006 5:06 pm

2 mangle rules, one for packets going TO your Asterisk server, one for packets coming FROM your Asterisk server. No need to worry about VOIP. Just make sure you have set up one main queue for each interface, and 2 (or more) queues - one for VOIP, one for everything else. Set your everything else limit way below 5.5Mbps, maybe 1.5Mbps, and use bursting to enable faster speeds. Set no limit on VOIP, and a max limit on your interface at around 4-5Mbps.
Are you sure this is correct? My understanding of the workings of the queue tree would suggest that in the setup you describe above you would service the 1.5Mbps "everything else" traffic before "VoIP" as "everything else" would be a leaf but VoIP would always be of the parent and therefore serviced later regardless of the priority assigned to the queue.
 
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HarvSki
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Thu Nov 09, 2006 10:51 am

You may have a "hidden node' type problem where one client is talking over another as it cannot hear all the other clients itself, CDMA. You can avoid it by setting the RTS of each client to say 256 (leave fragmentation alone and leave RTS settings on the AP alone too), this will force the client to ask the AP if it is OK to talk. Interestingly if you use MT as client there is not RTS option! If you were to use MT everywhere then you could use NStream protocol which would also avoid the hidden nod problem.
 
transporter_ii
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Re: STRANGE VOIP PERFORMANCE

Tue Jan 23, 2007 1:44 am

We currently have around 30 clients using the system all sharing 4Mbit connection and limited to 128k/256k or less
I have recently set up an asterisk box to be able to give voip service to our clients.

Thanks
David
This is an old post and someone has already pointed out that 30 clients on 1 radio is too much...but I thought I would add a note here for people doing research. With Cisco's high-dollar APs they recommend (and you can find this at Cisco.com by searching google):

The number of 802.11b phones you can deploy per Layer-2 subnet or VLAN depends on the following factors:

•No more than seven G.711 or eight G.729 active calls per AP

Also, Network World did some actual testing on some super-high dollar APs optimized for VoIP traffic, and running through a special switch that cost many thousands of dollars. They topped out at 11 or 12 concurrent VoIP calls on 802.11b, and reported problems if they went one over that number. Sorry I can't remember more details, but after reading the article, and Cisco's recommendation, I figure 8 - 12 concurrent calls is about the limit for 802.11b...on good quality equipment.

Just my two cents.
 
ldvaden
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Tue Jan 23, 2007 6:51 am

We used to use 15db omnis, and now use 16db sector panels, and find that the sectors have way better coverage area over the omni.
Please tell us which manufacturer makes an Honest To God 16 dBi gain sector panel antenna?

regards/ldv

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