Factors affecting VoIP quality
There are several factors that profoundly impact the quality of voice over the Internet. These factors can be described in terms of their general affect on VoIP quality: Negative or Positive.
There are several factors that profoundly impact the quality of voice over the Internet. These factors can be described in terms of their general affect on VoIP quality: Negative or Positive.
Negative Factors
Of the three negative factors for VoIP performance, the first one is delay, which results in echo and talker overlap. The second one is jitter, which is essentially the variation in delay. The third problem is packet loss. These factors are explained in much more detail below.
DelayDelay results in echo and talker overlap. Echo becomes a problem when the round-trip delay becomes high. Talker overlap (the problem of one caller stepping over the other talker’s speech) becomes significant if the one-way delay becomes greater than 250 milliseconds.
JitterJitter is essentially the variation in delay. This is primarily introduced because of the variation in inter-packet arrival time.
Of the three negative factors for VoIP performance, the first one is delay, which results in echo and talker overlap. The second one is jitter, which is essentially the variation in delay. The third problem is packet loss. These factors are explained in much more detail below.
DelayDelay results in echo and talker overlap. Echo becomes a problem when the round-trip delay becomes high. Talker overlap (the problem of one caller stepping over the other talker’s speech) becomes significant if the one-way delay becomes greater than 250 milliseconds.
JitterJitter is essentially the variation in delay. This is primarily introduced because of the variation in inter-packet arrival time.
Packet Loss
Packet loss is a constant problem in packet-based networks. In a circuit-switched network, all speech in a given conversation follows the same path and is received in the order in which it is transmitted. If something is lost, the cause is a fault rather than an inherent characteristic of the system.
Apart from these factors there could be impairments caused by codecs. These impairments are due to the distortion introduced by the codec and due to the interaction of network effects with codec operation. Speech coding and compression Both speech coding and compression have been used in the traditional telephony for over two decades. With the exception of the local loop, almost all voice is carried over the PSTN in digital format. The received analog voice undergoes an analog-digital conversion at 8000 samples per second with 8 bits per sample, producing a 64 kbps digital data stream. A codec is the device that performs the conversion from analog voice into a digital format and vice-versa. The standard method used in traditional telephony is PCM (pulse code modulation) implemented by using a codec that conforms to ITU-T standard G.711. Most humans can hear sound up to about 20 KHz, but the traditional telephony uses low-pass filtering to remove everything but approximately the lower 4 KHz of the speech signal. In addition to this, voice over packet networks commonly use low bit rate codecs for compressing the received noise. These low bit rate codecs preserve the parts of the speech that are of important to the human listener taking out those parts that are not of any importance such as silence, redundantly long words. This is generally known as perceptual coding and is used in a number of other areas too, such as MPEG-2 video compression, JPEG image compression and MP3 audio. Standardized codecs have been tested with multiple speakers and multiple languages. The results can be tabulated as below.
Packet loss is a constant problem in packet-based networks. In a circuit-switched network, all speech in a given conversation follows the same path and is received in the order in which it is transmitted. If something is lost, the cause is a fault rather than an inherent characteristic of the system.
Apart from these factors there could be impairments caused by codecs. These impairments are due to the distortion introduced by the codec and due to the interaction of network effects with codec operation. Speech coding and compression Both speech coding and compression have been used in the traditional telephony for over two decades. With the exception of the local loop, almost all voice is carried over the PSTN in digital format. The received analog voice undergoes an analog-digital conversion at 8000 samples per second with 8 bits per sample, producing a 64 kbps digital data stream. A codec is the device that performs the conversion from analog voice into a digital format and vice-versa. The standard method used in traditional telephony is PCM (pulse code modulation) implemented by using a codec that conforms to ITU-T standard G.711. Most humans can hear sound up to about 20 KHz, but the traditional telephony uses low-pass filtering to remove everything but approximately the lower 4 KHz of the speech signal. In addition to this, voice over packet networks commonly use low bit rate codecs for compressing the received noise. These low bit rate codecs preserve the parts of the speech that are of important to the human listener taking out those parts that are not of any importance such as silence, redundantly long words. This is generally known as perceptual coding and is used in a number of other areas too, such as MPEG-2 video compression, JPEG image compression and MP3 audio. Standardized codecs have been tested with multiple speakers and multiple languages. The results can be tabulated as below.
Here MOS is the measurement for voice clarity. This is explained in detail later in this chapter.Positive Factors
Of the two positive factors for VoIP performance, the first one is bandwidth, which is absolutely necessary for adequate performance. The second factor is prioritization. Prioritization becomes increasingly important as the network gets congested.
Of the two positive factors for VoIP performance, the first one is bandwidth, which is absolutely necessary for adequate performance. The second factor is prioritization. Prioritization becomes increasingly important as the network gets congested.
Bandwidth
One of the greatest challenges of VoIP is voice quality, and one of the keys to improving voice quality to an acceptable level is bandwidth. Therefore, additional bandwidth is certainly needed, if only to support additional traffic as demand for VoIP continues to grow. While additional bandwidth is a necessity for a network that is required to support voice in addition to the data traffic that is might have carried all along, additional bandwidth is not, by itself, a complete solution for the QoS issue.
PrioritizationOnce sufficient bandwidth is available to enable high-quality voice transfer, we need to control and prioritize access to the available bandwidth. As this regulation is not exerted over the Internet presently and because the IP is designed completely for the transfer of the data from its outset, depending upon the usage of the bandwidth, voice quality over the Internet might vary from acceptable to atrocious.
PrioritizationOnce sufficient bandwidth is available to enable high-quality voice transfer, we need to control and prioritize access to the available bandwidth. As this regulation is not exerted over the Internet presently and because the IP is designed completely for the transfer of the data from its outset, depending upon the usage of the bandwidth, voice quality over the Internet might vary from acceptable to atrocious.
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