How should an enterprise test its network to prepare for VoIP?

We conduct network voice readiness assessments prior to deploying VOIP, especially on large installations that span a WAN with several sites. Voice is delivered as UDP traffic, so you only get one chance to get it there on time. No retransmit feasible. Jitter, packet loss, and round trip delay all affect your voice quality. QOS and prioritization for voice in your network is critical. Implementing a voice VLAN is also quite helpful. You can have a screaming fast 10G network, but if you can't reliably deliver 64K voice packets, you will have the equivalent of bad cell phone voice quality. You can't just treat your network as a data network anymore. It is now a converged voice and data network. Time, money and effort invested in a network voice readiness assessment is money well spent in the long run. To quote a colleague of mine: "voice is the canary in the data network". Miners use to take canaries in to the mine with them. If the canary died, it was time for them to get out. If you have problems in your data network, you will notice it in your voice traffic. So spend the time upfront, test your converged network, and get the network "voice ready" prior to deploying VOIP.

This is a re-post answer which was modified to fit this blog. I've deployed over 30 managed sites within the past year alone. Managed in the sense that - my company was tasked with providing everything from the ground up on most occasions including provisioning the connectivity. So what was involved with deploying successful VoIP infrastructures and what do I look for first-hand.

On an optimal design, I NEVER want voice and data (data meaning email, http, etc) running on the same network whenever possible. Because VoIP creates its own data (SIP, invites, byes, etc.) which is rather chatty, I've always seen better results keeping things segregated. This also helps in a few different ways: 1) If data becomes a bottleneck (downloads, etc), call quality isn't affected. 2) Security: It is a little more layered. If you have port-security even better since it would be difficult for someone to jump on a VoIP only network/switch with VLAN's + port-security which would allow them to eavesdrop. Also, because many IP Phones have browser capabilities, with the right http tampering, an attacker can create some nifty POSTs and GETs which can reboot phones, reconfigures phones, eavesdrop on voicemails and the list goes on.

The issue surrounding VoIP that many don't get is - and always will be - speed period. Many also tend to forget that VoIP is almost always going to go over an unstable network - whether one likes it or not. Unstable being that - no matter what you *think* you can do - networks will fail and almost always do even under the strangest circumstances (http://tinyurl.com/na9ly5) With this said, I'd also like to point out that QoS is not really a factor in fact, those promoting QoS in my opinion need to learn a lot more about networking before solely stating "QoS!" ( http://www.bricklin.com/qos.htm ) Unless you're planning on having point to points at all locations (very difficult for the SoHo to MidSized companies), QoS means nothing. Just because you paint a packet doesn't mean another carrier has to honor it.

Some of the things you should do is go over your design with a fine tooth comb. Make a solid determination of how much bandwidth your devices will use based on what CODECS you will be implementing, what kind of equipment you have and what are their capabilities. What kind of switches perform at what rate, how fast, how often. Don't skimp on spending a little more. Far too often I've seen people cut corners and opt for say, a smaller switch. Invest properly!

If you're going to be implementing VoIP in your existing network, the best method to get an accurate view of how it will perform would be to perform a network analysis over the course of about 10 business days. What you want to look for is bandwidth rates. You want to make sure you're always going to be under at least 65% utilization WITH your new VoIP infrastructure. This analysis can be done with tools like Wild Packets Omnipeek, Wireshark, etc., you want to see what is going on - on a daily basis. How much traffic is being used, how its being used, what kind of performance/errors are being seen.

For example, supposing you purchased all high end equipment from a vendor - it could be the top of the line. If mis-configured, you're likely to get a lot of errors, packet loss, etc., in the end your VoIP MOS score will be horrible. It all needs to play well together so even a fast network of say 10G speeds, CDP running could leave your call sounding as if someone where on a helicopter (stutter). You could try call generators but the honest truth is, they'll never even come close period. Best bet to test robustness, set your network up to allow you to FTP large files (bandwidth generation) and test REAL calls during this time. See how your network REALLY holds up. This is a more realistic gauge than relying on too much automation.

Finally (pretty long no?) your network provider. If you can find me one right now that offers immaculate uptime, I'll buy you beer for the rest of the decade. Because I'm on the managed realm at the carrier level, there is hardly a provider I haven't used or dealt with. Level3, Global Crossing, XO, Paetec, Covad, Verizon, AT&T, etc., you name them, I've likely dealt with them. As stated and linked, (NANOG) networks will be networks. You're best bet is to be prepared in your design. Because you're doing VoIP, at some point in time you will either need a PSTN to VoIP encoder/decoder slash Session Border Controller or other device to perform the conversion, or you will seek out a ITSP (Internet Telephony Service Provider) to do it for you. You're now at the mercy of a network outside of your control. Redundancy can also be challenging in this environment, remember an Internet hiccup has the potential to knock your entire service offline. Even if its for say 5 seconds, those can potentially be thousands of dropped calls. So design, design, design... Have someone who understands network analysis take a good view and create a thorough design based on what's visible and DO NOT SKIMP ON BANDWIDTH. Besides, bandwidth is dirt cheap right now.

Whoever is providing the VoIP solution should be ensuring your service provider, network and policies are set up to handle and prioritize traffic for voice. Bandwidth, latency, jitter and packet loss all need to be considered when determining readiness and an Internet MOS score can help. Get rid of hubs and cheap routers as they can create packet collision and destroying voice quality. Voice is quality sensitive and garbled audio and chopping will occur when poor quality and low bandwidth are present, where data is not sensitive and pages will just load slowly. We use a tool for our customer deployments that measures our traffic and gives a MOS score for each network you run it from. http://www.worldsmartcentral.com/test/

PanTerra Networks offers WorldSmart, a unified communications software-as-a-service (SaaS) which includes a hosted VoIP PBX with ACD and unlimited calling, electronic fax, instant message, mobile text, email, calendar and contact management all with presence unified through a single administrative and user interface. Unified Communications at the next level based also includes desk-top sharing, web collaboration, and conferencing, all services that PanTerra has integrated seamlessly with WorldSmart. Dave Immethun http://www.panterranetworks.com