Cisco SPA962-NA Administration Guide - Page 19

Chapter 1 Introducing Linksys 900 Series IP Phones SPA900 Series Features Note SPA900 Series IP phones support all the above voice coding algorithms. The following factors contribute to voice quality: • Audio compression algorithm-Speech signals are sampled, quantized, and compressed before they are packetized and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with 12-16 bits per sample. The compression algorithm plays a large role in determining the voice quality of the reconstructed speech signal at the other end. SPA900 Series IP phones support the most popular audio compression algorithms for IP Telephony: G.711 a-law and µ-law, G.726, G.729a, and G.723.1. The encoder and decoder pair in a compression algorithm is known as a codec. The compression ratio of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate, the smaller the bandwidth required to transmit the audio packets. Although voice quality is usually lower with a lower bit rate, it is usually higher as the complexity of the codec gets higher at the same bit rate. • Silence suppression-SPA900 Series IP phones apply silence suppression so that silence packets are not sent to the other end to conserve more transmission bandwidth. Instead, a noise level measurement can be sent periodically during silence suppressed intervals so that the other end can generate artificial comfort noise that mimics the noise at the other end (using a CNG or comfort noise generator). • Packet loss-Audio packets are transported by UDP, which does not guarantee the delivery of the packets. Packets may be lost or contain errors that can lead to audio sample drop-outs and distortions and lower the perceived voice quality. SPA900 Series IP phones apply an error concealment algorithm to alleviate the effect of packet loss. • Network jitter-The IP network can induce varying delay of received packets. The RTP receiver in SPA900 Series IP phones keeps a reserve of samples to absorb the network jitter, instead of playing out all the samples as soon as they arrive. This reserve is known as a jitter buffer. The bigger the jitter buffer, the more jitter it can absorb, but this also introduces bigger delay. Therefore, the jitter buffer size should be kept to a relatively small size whenever possible. If jitter buffer size is too small, many late packets may be considered as lost and thus lowers the voice quality. SPA900 Series IP phones dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call. • Echo-Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. SPA900 Series IP phones have a near-end echo canceller with at least 8 ms tail length to compensate for impedance match. SPA900 Series IP phones implement an echo suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable. • Hardware noise-Certain levels of noise can be coupled into the conversational audio signals because of the hardware design. The source can be ambient noise or 60 Hz noise from the power adaptor. The SPA900 Series hardware design minimizes noise coupling. • End-to-end delay-End-to-end delay does not affect voice quality directly but is an important factor in determining whether subscribers can interact normally in a conversation taking place over an IP network. A reasonable delay figure should be about 50-100 ms. End-to-end delay larger than 300 ms is unacceptable to most callers. SPA900 Series IP phones support end-to-end delays well within acceptable thresholds. Document Version 3.0 Linksys 900 Series IP Phone Administrator Guide 1-5