Factor Affecting Voice Quality in VOIP Calls

A VoIP system is regularly used in homes and business around the globe. The system is perfect for making phone calls with extended features and a more affordable price, but many times, users do not have the best quality in a call. It is important to make sure that the call quality is efficient for your needs. VoIP can be used in many instances, from emergency communications, to long-distance only, etc. Evaluating your VoIP need, will help to determine the voice quality needed.

VoIP offers a range of different features to users. One of the most important features is video conferencing. Through video conferencing, you will be able to save a lot of money when it comes to travel expenses and traveling time.

In a business environment, VoIP can be used on desk tops, on IP phones, or on a hand held phone using a wireless system. This brings a new dimension to the game. Would you have the same voice quality all over your office? Will you be able to send an important file to the client when you speaking to him using your tablet?

Take a look at the factors that affect your connectivity and transmission quality. Keeping these under control will help you have a good and robust IP communication system in place.


Your bandwidth is often mistakenly thought of as your connection speed. In general, connection speed is measures as bits per second. This could be thousand (kilo) bits per second (Kbps) or million (mega) bits per second (Mbps).

Bandwidth, on the other hand comes from radio or electronic transmission. It is the width of the range (or band) of frequencies that an electronic signal uses on a given transmission. Bandwidth is the difference between the lowest frequency and the highest frequency. Since frequency is measures in hertz, bandwidth is the difference between the lowest and highest frequencies. A typical voice signal has a bandwidth of 3000 hertz, or 3 kHz.

VoIP needs two protocols – a signaling protocol such as SIP, H.323, or MGCP that is used to set up, disconnect, and control the calls. A second protocol is needed to carry voice packets. Most VoIP systems use Real Time Transport Protocol or RTP.

An IP phone generates a voice packet every 10, 20, 30, or 40 millisecond, depending upon implementation.  To calculate the bandwidth you need to understand the following:

  • Packet size for voice (10 to 320 bytes of digital voice)
  • Codec and compression techniques
  • Header Compression
  • Layer 2 protocols used
  • Silence/voice activity detection

Depending upon all these factors, the bandwidth requirements vary from 19.6 Kbps to 87.2 Kbps. A good rule of thumb is to reserve 24 Kbps of IP network bandwidth for each call.

Connection Types

While connection speed is related to bandwidth, it will refer specifically to the speed at which data is transferred through a single point. Connection speed is measured in bits per second. Certain types of connection such as dial-up are often too slow for you to be able to use VoIP. The only VoIP-related feature that you will be able to use in this situation is instant messaging, a type of feature that existed even during the days of dial-up Internet.


Do not forget that you need to have the appropriate equipment as well, be it the router or the IP Phone that you use. When you are procuring equipment for VoIP, always get as much information as you can about the ATA, router, IP Phone, switches, etc.

Packet Loss

As we discussed before voice is sent as compressed data packets. Sometime, packets can be lost on the way. Unlike real data, where the receiving computer can ask for a retransmission, dropped voice pockets are lost forever. This will mean loss of conversation.


Latency refers to the time delay caused by network traffic, transmission delays, and processing delays. When this happens, there will be a delay between the time when you spoke and when it is received at the other end. When one set of voice data packet is delayed and next one reaches faster, what you will hear is a garbled message.


Jitter happens when two set of sequential voice data packets dot not travel at the same speed or with the same time difference. This is also called variable latency between packets. Jitter is common in IP based systems because the route chosen for multiple packets may always not be the same

Michelle Patterson blogs on technology extensively. She understands and writes about IP/VoIP and Unified Communication. She works with a few leading companies to understand emerging technology, and transfer that knowledge to rest of the world.

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