Adding VoIP to an existing network

Adding VoIP to an existing network

ITKnowledge Exchange member "bklynbound" needed to know if he could implement a new VoIP system, and fellow techies jumped in to help out. But could he keep costs down and use the company's current hardware? Here is a portion of the conversation. You can read the rest of the thread on ITKnowledge Exchange.

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ITKnowledge Exchange member "BKLYNBOUND" asked:
I have been tasked with finding a way to save money in our telecommunications costs. VoIP can save us money in the long run, but do I risk headaches by implementing a new system on our current hardware? Is there a VoIP solution that will let me keep my existing Avaya Partner (PBX?) Key System- ACS R7 in our Los Angeles office and our Toshiba Strata 280 Key System here in New York? We have a SDSL 1.5 MBPS connection to the net in New York and a SDSL connection in Los Angeles as well. The offices are connected through a VPN tunnel between two netscreen devices.

"JRMOORE" WRITES:
Unless you are hell-bent to suffer the pains of pioneering or want to spend a lot of bucks with Cisco, I would suggest contacting COVAD for your primary offices in New York and Los Angeles. They have a backbone of their own and much higher quality of service than the other BrandX guys. There is probably a 70% chance you can save money and even keep your vintage PBX system with their offering of PBXi. Then you can use that until you feel it valuable to go straight VoIP with new handsets. Good luck!


"HEDGEHOG" WRITES:
You should analyze your phone bill to see the percentage of intra-office calls. You will be surprised to discover that a large number of calls are between your offices.

If that is the case and you have a semi-decent broadband connection on both offices, I suggest you install out-of-the-box VoIP gateways to get free calls between your offices. You should look for one that offers you the option to plug in directly into your existing PBX. They appear as additional extensions and are automatically routed to the remote office. They are fairly simple to set up, you get good support from the vendor or reseller and get to keep your existing kit for a few more years until you decide to go all-out for a VoIP PBX and sign up for VoIP providers as JRMOORE suggested.

Check out the MultiVoIP gateways from the good ol' Multitech people.

Another alternative is to set up SIP proxies on each end and use either softphones on the PCs or hardware VoIP phones in your LAN. There's obviously more configuration for you to do, but SIP is a recognized standard (as opposed to Skype!), there are several open-source SIP proxies and I am sure there must be some SIP registrar in your area that will provide SIP services and numbers.

Check out siproxd for FreeBSD, Linux, OpenBSD, OS X and Solaris. We use it here and it works great. .

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"PAUL144HART" WRITES:
The idea of using SIP/VoIP on your data network needs to be considered carefully -- you'd want to define a QOS VLAN for the VoIP so the calls don't get packets lost due to congestion. Using the old PBXs like that would require a gateway such as the Cisco/Nortel. You might also need an Adtran or other box to bring the T1s to the PBX -- you can split a T1 easily and allocate numbers of ports to your telephone provider, depending on your site size and configuration. Hopefully the PBX does have T1 open spots! I think I've seen adding a T1 card to a PBX to be approximately $3000 (was an old Nortel PBX). The open source proxies are okay -- but you don't get technical support. There are quite a few companies out there that would give you a free quote if you want a jump on what it may cost.

For analyzing the cost you may want someone to do the financial judgments -- you'll probably want either internal rate of return or a break-even point to show that it actually will save money. Someone with an MBA...


"PAUL144HART" WRITES:
For your netscreen boxes, make sure they will allow the SIP or H.323 protocol and RTP packets.

With 1.5M of SDSL, you could allocate about 200K of bandwidth for the VoIP traffic -- many due to the fact you have a T1 PRI trunk. Not said if you are using all of that for calls.

You can put in a couple softphones and a proxy to test connectivity out for not much money. Better SIP softphones are eConf or Pingtel. If you look at H323, there is free softphone software founded by Pingtel.

Remember, Real-Time Transport Protocol (RTP) streams are 8KHz/8bit or 16KHz. Each stream is a voice channel.


"MIKEL69" WRITES:
I would recommend spending the bucks with Cisco, but of course I am biased. But this will offer the most flexibility/compatibility with your current infrastructure.


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    This was first published in March 2006