Tips to a VoIP Friendly Network

Deploying a VoIP network can be full of surprises to the novice and unprepared. Knowing the basics of what your phone calls require can be the difference between a crystal clear calls experience or a seemingly never ending flood of support issues. We have compiled the following reference to help you through the process.


Cable/DSL Modem - Includes a “key” to access your Internet Service Provider’s (ISP) network.

Router - Routes traffic to several devices, often includes wireless capabilities.

Switch - Creates network ports, POE (Power over Ethernet) and manages VLAN assignment for your network. 

Lets start with some basics:

VoIP calls pass as data from your phone to your phone server (or cloud service) and utilize your company network and internet services to route calls. Each component of your network plays a part in this process and each can have capabilities that help optimize the quality of your voice. There are two major components that drive the quality of your voice.  First is latency or jitter, you can think of this as the consistency of the speed of voice packets that arrive at the destination. If some packets are arriving late they will be excluded from the resulting voice stream that is being played. The second is packet loss, if any of the packets are being discarded this will impact the resulting audio quality. If you are experiencing audio issues one or several of these legs could be a contributing factor. It is important to test and isolate individual components in your network to come to a resolution the quickest. The nice thing is if you take the time to make sure you are operating on a firm foundation this will be a one time exercise that will provide quality phone calls for a very long time.

Recommended Modem Configuration

Sometimes a combo device ( gateway) is provided by your ISP, which includes a modem and a wireless router.  The router in a combo-device is only (a) for internet browsing and email or (b) to support the ISP’s proprietary phone service.  Gateways are not friendly to VoIP, as they do not translate the data correctly; however, one of these two changes can easily be made:

Bridge the Gateway - This “builds a bridge” over the routing capabilities of the combo device allowing data to pass through without [incorrect] translation.

Replace the gateway with a “dumb modem” - a modem with NO routing capabilities

Essentially, both methods are a way to setup the modem/ gateway to stop trying to translate VoIP data.  This, instead, allows the modem to pass the original data to a router that can then translate the VoIP data properly.

Recommended Router Configuration

Your router plays one of the most critical roles in prioritizing and routing your VoIP phone calls. If your internet service provider is providing you a router chances are it will not have the facilities to optimize any other VoIP services but theirs. If you find this is your case simply have them setup the modem in pass-thru or bridge mode and implement your own VoIP friendly router. You should look for a router that supports QOS (Quality of Service) and if you have 50 or more extensions you should also consider having your phones on their own VLAN.

QOS (Quality of Service) - Your phone system places a tag on all of the packets that contain voice. Your router can be configured to automatically send these packets before it would send a standard email or other data from your network. This can drastically improve audio quality.

VLAN - VLAN's help you to isolate your voice network for your computer network by placing a VLANID tag on packets from your phone network. Isolating the network adds another dimension to you capabilities to apply QOS.

Security (Don't Skip This Part)

Since the beginning of phone networks people have been conspiring sophisticated hacks to place free phone calls. VoIP is no exception. Your firewall is a critical component to allowing the good guys in and keeping the bad guys out. If you are using a cloud solution or do not have any phones outside your internal network you are in good shape and you can completely lock down your network from uninvited visitors. In most cases your default firewall settings are adequate. The rest of this applies to premise phone systems with remote devices or a direct IP based VoIP account with a VoIP provider. You will need to open ports for these devices IP's to allow them to access your phone system. The most secure way to do this is to use an IP based access rule that maps the needed ports to your telephone systems IP address. You'll need to know the IP addresses of your remote devices and add a specific rule that maps port 5060 UDP and 10000-20000 UDP to the phone systems internal IP. Lastly if you have remote phones who’s IP addresses may change (smart phones, home cable modems) you will need to allow all IP addresses access to those same ports. This is by far the most insecure method and relies on the phone systems capabilities to identify and block hacking attempts. Using a VPN (Virtual Private Network) to connect remote offices will add a layer of encryption and enable you to further secure your network.

Recommended Switch Configuration

There are three important components to consider when purchasing your network switches:

Managed vs Unmanaged - Managed switches will give you a lot more options and reporting capabilities to configuring your network. You should always choose a managed switch when purchasing a new switch for a VoIP Network. 

POE - (Power Over Ethernet) - POE will simplify your desk phone deployment and give you the capabilities to implement a battery backup solution that enables your phone calls to continue when the power goes out. POE provides power to your phones through the existing network cables eliminating the need for the phones power brick.

VLAN - Larger networks can isolate their VoIP devices into separate VLANs. VLANs will enable you to establish advanced routing and security options for your voice traffic.

BANDWIDTH (speed of your the network)

You’ll hear, or have heard, the terms “upload and download speed”.  This is the bandwidth, or speed, of your network.  It is very important and seldom changes.  The rule of thumb is simple:  the more users you have, the faster speed you need.  Without a fast enough speed, only one or two people may be on a call at the same time or the quality of the calls will be bad.  With enough speed, you’re in a good place for all users to be on the phone.

STABILITY (quality of your network)

The biggest causes of network instability are packet-loss and latency.  We’ll call this “incomplete data”.  It’s not uncommon to have the fastest bandwidth available and still have a high amount of instability. 

The instability is caused by packet-loss. Imagine “packets” as packages on an open-top delivery truck in route from New York to Los Angeles. If there is instability on that road, those packages are likely to fall off the truck. By the time the shipment (data) arrives in LA, it’s incomplete. This is no different with data. For voice calls, you need the conversation (data) to be received in completion.

A VoIP friendly network should have 0% packet loss.
Instability also leads to latency. Imagine there’s traffic on the road and packages aren’t delivered on time.  Internet VoIP data works the same way.  The only difference?  There’s no HOV or speedy tolls lanes in the internet.  Everyone gets equal priority to use the road and, as a result, the latency in sending VoIP data between two points can vary significantly.  What complicates matters?  If two packages are sent at the same time and they arrive at your door out of order, it’s not a big deal.  BUT!  If you speak two words and they arrive out of order?  Well, that’s a big problem. Yoda can be fun to imitate but may hinder your business.  A VoIP friendly network should have < 100ms latency 
These types of instability may happen once a week, but sometimes network connections are plagued with occurrences several times throughout the day.  For most internet functions such as email, browsing, and video streaming, you aren’t likely to see many issues with this instability. How and why, you ask?
You type “” into your web browser.  Whether it loads immediately or takes 15 seconds, the page will eventually become complete.  An image may take awhile to load, but the webpage will appear soon enough, as will the rest of the page.

You go to YouTube and look at that funny video your coworkers are passing around, which is only 15 seconds long.  Though it may take ten seconds or an hour, that video will eventually fully load for your viewing pleasure
These scenarios can most certainly be annoying, but imagine if that page or video never finished loading?  These are not “realtime situations”.  That web page will finally load, that email will finally arrive, and that video will finally play all the way through.

A phone call?   A phone call is a real-time situation.  You talk and the other person responds.  If packet-loss or latency are in action, you’re losing bits and pieces, if not all,of the phone call.  Call quality issues will manifest as jitter (alien-like voice and crackle), static, delay, echo, one-way audio (“I can hear them, but they can’t hear me.”), dropped calls, inability to make or receive calls.


Provide your ISP with proof.  Calling and saying, “I have a problem.  Fix it.”, will rarely end with good results.  There are a few ways to get evidence showing these interruptions. A ping test is a common method.  “Ping” is the same term use in the game Ping-Pong - a signal is sent from your network and bounced back several times.  The tests provide results on the “health” of your network by gauging the length of time that passes (measured in “ms”).

A simple test through the website “”, which provides readings of jitter, ping, and packet-loss.

An extended test through a company called VoIPSpear ( - Their free accounts allow you to run the same ping test consistently on your network.  You can login and it will provide graphs showing the state of your network’s stability over the prior six hours.  (This will require your IP Address.  You can get this at easily.)

We have a free comprehensive  VoIP Quality Test utility on our website at:

A VoIP friendly network should have:

  • Ping < 100ms
  • Jitter < 10ms
  • Packet Loss < 0.1%
  • A VoIP “unfriendly” network would have:
  • Ping > 300ms
  • Jitter > 30ms
  • Packet Loss > 1%
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