Audio Compression Algorithm; Silence Suppression; Packet Loss; Network Jitter - Linksys SPA1001 Administrator User Manual

Ensuring 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. The Linksys ATA supports 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

The Linksys ATA applies 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 lowers the
perceived voice quality. The Linksys ATA applies an error concealment algorithm to alleviate the effect
of packet loss.

Network Jitter

The IP network can induce varying delay of the received packets. The RTP receiver in the Linksys ATA
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 the jitter buffer size is too small, late packets may be
considered lost and this lowers the voice quality. The Linksys ATA can 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. The Linksys ATA has a near-end echo canceller with at least 8 ms tail length to
compensate for impedance match. The Linksys ATA also implements an echo suppressor with comfort
noise generator (CNG) so that any residual echo is not noticeable.
Linksys ATA Administrator Guide
1-4
Chapter 1 Introducing Linksys Analog Telephone Adapters
Document Version 3.1