Packet Loss

Packet LossPacket 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 FactorsOf 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.