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DirectWireless
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VOIP performance and advice

Mon Feb 05, 2007 3:42 am

I've been off forums the past few months doing some wireless equipment research. I currently have an all-MT 802.11b network (some 2ghz-5mhz stuff now too, and a separate all-5ghz-10mhz network too). I've been playing with Canopy 5.7 to deploy a new business VOIP network, but Canopy Advantage is 14Mbps (and with a 50/50 ratio, only 7Mbps each way, at best case). Those #'s are user data throughput, not radio speeds.

Canopy is supposed to offer up to 26 simultaneous calls per AP @ g711 with no increase using g729. I need this same performance, except I also want to be able to transport a few megs of data as well, which Canopy @ 26 calls means no more data left. I've only tested with 12 simultaneous calls so far (8 port Audiocodes FXS, 4 Grandstream ATAs). So far it's been so-so, even with a reflector. Canopy's whitepaper claims that the actual radio performance goes down to a dismal 4.2Mbps when loaded with calls - not good at all.

So I'm looking back to see if I can do this with a Mikrotik solution. I have done PTP links with MT having dual 60Mbps links. I'd like to be able to have up to 8 clients on an AP, and 26 SIP calls going, and have a few megs (I'm thinking 5 or 6) left for data. These 8 clients likely would have 10 lines each, and the server will be programmed to limit simultaneous calls per sector to 25 or 26. I'd like to investigate some form of dual-radio PTMP bonded solution as well so I can put 2 sectors per direction, a 5.3 and 5.8 using those Commex P4 MiniITX boards with dual MPCI slots.

I need some real world advice from you guys out there who do VOIP over Mikrotik PTMP. Is this a do-able thing? I don't know how well a single client's VOIP packets aggregate together on NStreme (like 10 SIP calls, will that meld into big packets easily?).

I'd also consider for the dual-radio setup some routing tricks so that one radio is dedicated for upstream, and one radio is dedicated for downstream. I wonder if the half-duplex problem were eliminated (and NStreme polling solves the hidden node), if I could actually run dozens of voice calls PTMP. Really the only other thing that would help would be some form of sync on the tower but that's a later thing anyhow and would mainly only help for frequency re-use.

Any input or suggestions?
 
believewireless
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Mon Feb 05, 2007 8:18 pm

We had tons of problems with NStreme and VoIP. Mainly due to the high latency and jitter of NStreme. Turning NStreme off fixed everything. However, we still see issues where a customer with a poor link will cause everyone to drop packets. So for VoIP, you might want to look into another solution until Mikrotik fixes these problems.
 
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stephenpatrick
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Tue Feb 06, 2007 12:23 am

Nstreme and VOIP - or Nstreme and WDS and VOIP?
P2P or P2MP?
Low or high interference environment?

Would be interesting to hear more detail -

Regards

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DirectWireless
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Tue Feb 06, 2007 12:27 am

Well here's what I'm thinking:

6 AP's on the tower, with a 5.3 VPOL and a 5.8 HPOL 120 degree sector in each direction. Using 1 P4 MiniITX board per pair of radios, and NStreme w/Polling. Each client gets a dual-polarity dish, dual radio, with separate TX/RX. Using some routing and/or filtering tricks, the traffic would only flow one direction on each link. The killer for VOIP & wireless is the latency of half duplex - full duplex means each SIP stream has no more radio limitations of half duplex.

I'm thinking you could run the downstream signal in 5.3 turbo mode to help comply with the less power and provide customers with a fat internet downstream and run 5/10/20mhz upstream depending on your VOIP requirements. I think you could easily pull 50Mbps downstream in turbo mode, and 20Mbps upstream - that's fast enough to blow away most offerings on the market today, albeit needing 60mhz of bandwidth per sector although between 5.3 and 5.8 you have 3 5.3 turbo channels and 5 5.8 non-turbo channels. For less capacity you could use 10mhz channels each way and still get 10Mbps full duplex, with all of the benefits of dual-frequency like cellular uses.
 
believewireless
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Tue Feb 06, 2007 4:32 pm

NStreme, WDS and VoIP in a low noise PtMP environment. Just turning off NStreme fixed everything. As mentioned in another thread, we tried different framing policies and sizes and none of them fixed the issue.
 
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janisk
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Tue Feb 06, 2007 4:52 pm

nstream collects packets into larger packets and then sends them all in one packets, thats why latency increases.

you have to have really intense traffic to get most of nstream
 
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stephenpatrick
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Tue Feb 06, 2007 5:10 pm

NStreme, WDS and VoIP in a low noise PtMP environment.
Is there a P2MP problem with Nstreme and WDS, especially with VOIP?
Anyone else experienced this?

Regards

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believewireless
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Tue Feb 06, 2007 8:36 pm

The problem is the jitter and latency that gets worse as you add subscribers. I agree that large traffic should help this but most customers don't send/transmit large amounts of data, especially with VoIP. This is why it works for PtP but not PtMP.
 
DirectWireless
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Tue Feb 06, 2007 11:27 pm

But I'm looking at 8+ phones calls from up to 8 CPE's - plus data- so I could theoretically see where 8x64k could be a significant bit of small packets at 512k per node, 4Mbps for all 8 nodes making 64 phone calls total. If the packet aggregation worked optimally this arrangement should work well and offer decent latency.

The big problem I look to overcome by using dual radio PTMP is the constant switching of TX and RX - there will still be beacons & acks running, but one way only traffic should significantly increase throughput.

I want to try it out in a lab environment and see what happens. I plan to use fast machines so I won't be CPU limited.
 
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stephenpatrick
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Wed Feb 07, 2007 11:11 am

DirectWireless, what you are proposing would be VERY clever if it works, and if you do, please do post some results.
I'm sure a lot of people would like to hear more.

I thought Nstreme2 only worked in P2P, but you are proposing doing this at the network layer. the "level of improvement" over regular single-radio TDD would be interesting.

Regards

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jober
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Wed Feb 07, 2007 6:09 pm

I have gone with a little different approach.
Over the past year and a half I have been installing the rb532 with SR5 and SR2. The idea is to run VoIP traffic at 10mhz over the 5.8Ghz and let the web surfers, gamers, email and down loaders go over the 2.4ghz link.
On the customers that don't have nor want VoIP from us I just turn them into a hotspot to help light up the town.
On the tower I have dual frequency antennas 5.8/2.4. I am going to add 5.3 or 5.4 (if it's ever workable) to move the customers off the 2.4 so that it's only for the hotspot service.
although the lower 5.X should be for the VoIP, the 5.8 for the http,email,etc and the 2.4 for hotspot. But with that setup I have to run 3 radios per board and the rb532 only has 2 ports. Or I have to add and extra AP bridge.
If I go with the add on AP I think I may try some mimo units.
 
mperdue
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Wed Feb 07, 2007 6:20 pm

I am running 5.8ghz links to each tower, then 2.4ghz relay down to clients. I have one tower with 60 cpe's and 28 sip telephones tied into a voice server back at my office with an outgoing pri for local calls and the rest are sent across the internet to an LD provider. Rarely do I have problems, mabye once every month I need to reboot the voice server itself. I'm feeding with a 3Meg internet connection.
 
DirectWireless
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Wed Feb 07, 2007 7:09 pm

Wow is that 2.4ghz on one AP or multiple APs? 60 clients in itself is impressive. Is it Mikrotik CPE's or regular 802.11b/g stuff?
 
mperdue
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Wed Feb 07, 2007 8:22 pm

The one specfic tower has one RB532 with two Sanyo 400mw a/b/g/ card. One is set to 5.8ghz and is the backbone link. The other card is 802.11b (not g). I use RB112's with a single Sanyo Card at each client location. Then I go to sipura voip units. I am running g.711U on most all except a few people that I"m testing g.729. I am even sending fax's with fairly good results. Fax is about 80% effective, in other words 20% of the time it dosn't go though and they have to resend. I am also running a packetter at the front end of the network that sets priorty for voip. My clients are still able to do speed tests and pull 1.15 meg connections.
 
jober
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Thu Feb 08, 2007 1:50 am

Is that Sanyo card a Senao with a Sanyo sticker on it?
 
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ivaring
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Thu Feb 08, 2007 5:58 pm

I am running 5.8ghz links to each tower, then 2.4ghz relay down to clients. I have one tower with 60 cpe's and 28 sip telephones tied into a voice server back at my office with an outgoing pri for local calls and the rest are sent across the internet to an LD provider. Rarely do I have problems, mabye once every month I need to reboot the voice server itself. I'm feeding with a 3Meg internet connection.
Well, mperdue (it sounds like italian)

Congrats.

One question: regarding your customers, all of them are down last MTK in a wired structure?, or a wireless?.

Thanks.
 
mperdue
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Fri Feb 09, 2007 5:59 pm

My most recent clients are all running rb112, my early customers are running Linksys routers (loaded with sveasoft code). I no longer use the Llinksys/sveasoft due to the issue of being very difficult to get the right version of the linksys unit that you can load the svasoft code on. Linksys keeps chaning model names, and internal code and versions.

I'm in Virginia, U.S.

I'm using all Senao NMP-8602 cards. Though I do have one Rb532 that has a rebooting problem that I'm working on in another discussing. But thats the only unit that I have three cards runing.

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