Common cause of poor VoIP Quality and Solutions

How much of your current bandwidth is needed for high-quality voice calls?

The bandwidth requires depends on codecs compression and the number of concurrent calls you want to make. The table below shows the minimum bandwidth required to make calls  as well as recommended speeds for optimal performance.

Concurrent Calls  Minimum Required Bandwidth Recommended Bandwidth
1 100 Kbps Up and Down 3 Mbps Up and Down
3 300 Kbps Up and Down 3 Mbps Up and Down
5 500 Kbps Up and Down 5 Mbps Up and Down

Your home or business Internet  is also being used for other functions (web browsing, sending and receiving email, file transfers, etc) that are candidates contending for bandwidth.  Some but not all routers have the ability to prioritize voice services so that the impact of other applications doesn’t degrade voice quality.  This feature is also know as Quality of Service “QOS”.  It helps  prevent audio issues caused by voice and data competing for the same bandwidth.

Make sure your network router’s Quality of Service (QoS) settings are set as follows

  • UDP/5060 – Priority: High
  • UDP/5160 – Priority: High
  • UDP/16384 to 32768 – Priority: High

Some router function that we recommend to disable to improve the Voip Quality

  • Application Layer Gateway “ALG”
  • Stateful Packet Inspection “SPI”

Jitter is a common problem of the connection-less networks or packet switched networks. Because the information (voice packets) is divided into packets, each packet can travel by a different path from the sender to the receiver. When packets arrive at their intended destination in a different order then they were originally sent, the result is a call with poor or scrambled audio.  The maximum allowable duration of jitter is less than 10 ms before deterioration occurs.

There are 3 types of delays;

  1. Propagation Delay: Light travels through a vacuum at a speed of 186,000 miles per second, and electrons travel through copper or fiber at approximately 125, 000 miles per second. A fiber network stretching halfway around the world (13, 000 miles) induces a one-way delay of about 70 milliseconds (70 ms). Although this delay is almost imperceptible to the human ear, propagation delays in conjunction with handling delays can cause noticeable speech degradation.
  2. Handling Delay: Devices that forward the frame through the network cause handling delay. Handling delays can impact traditional phone networks, but these delays are a larger issue in packetized environments.
  3. Queuing Delay: When packets are held in a queue because of congestion on an outbound interface, the result is queuing delay. Queuing delay occurs when more packets are sent out than the interface can handle at a given interval.

The maximum duration of latency that a VoIP system can sustain without deterioration of service is less than 150 ms in any one direction.

Most ISP’s are designed for web surfing and not VoIP advantages. Transporting voice packets is different and requires an additional set of internet protocols that your ISP may not be providing.

Solution: Internet providers offer Business Class High Speed and DSL internet service that are acceptable.

Not all router are made equal.  Bad equipment is bad equipment.

Solution: Install a Specialized VoIP Router that has the ability to prioritize VoIP traffic.

If your company decides to route both voice and data over the same network without properly configuring your network for VoIP traffic, you can expect to have call quality issues.

Solution: This is one of the easiest and least expensive problems to correct. A Business VoIP capable router that is properly configured will generally solve the problem.

Live Outage Map by Downdetector

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Additional VoIP Testing with MOS

 Application such as VoIP will fail if the network quality is poor. The very nature of audio demands that packets flow through the network consistently, failure to do so causes the audio quality to drop.  Most users and providers are fixated of bandwidth speed however bandwidth is seldom the issue, quality and consistency of the packets are by far the most significant threat to VoIP.

Interpreting Results

The VoIP Speed Test will show you Jitter, Packet Loss, and MOS.  You should run this test multiple times throughout a normal work day to see if there is any variance in MOS based on the time of day.  For instance, during the peak working hours between 7am and 9am where user are logging in to the network.

A variation in packet transit delay caused by queuing, congestion, timing drifts, route changes and serialization effects on the path through the network; the maximum allowable duration for jitter is less than 10 ms before deterioration occurs.
When data packets get lost between point A and point B, this is called packet loss, and is measured as a percentage of overall packets. Ideally, you would want this to be 0%, however packet loss up to 1% is typically unnoticeable.  Most network that has packet loss are not noticeable until they implement VoIP because standard data streams (such as web surfing or watching videos) have the ability to re-send lost packets. VoIP on the other hand is a live conversation, so it ends up sounding like choppy audio or cut-outs.
The measure of time delay in moving packets from the transmitting agent to the receiving agent; the maximum duration of latency that a VoIP system can sustain without deterioration of service if less than 150 ms in any one direction.
Mean Opinion Score – Jitter, Packet Loss, and bandwidth are combined with the type of VoIP audio stream being used (called a codec) to calculate your MOS. A MOS of 4 to 5 is excellent. 3-4 is OK, but you can probably make some simple changes to improve. Anything below 3 means that your network is not ready for VoIP and needs some improvement before quality voice can be achieved.

Mathematically, 4.41 is the maximum MOS for G711, while the maximum is 4.07 for G729.

VoIPTestResult