© 2006 Cisco Systems, Inc. All rights reserved.ONT v Describe Cisco VoIP Implementations Digitizing and Packetizing Voice

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Basic Voice Encoding: Converting Analog to Digital

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Basic Voice Encoding: Converting Analog to Digital

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Analog-to-Digital Conversion Steps 1. Sample the analog signal. 2. Quantize the samples. 3. Encode the value into a binary expression. 4.(Optional) Compress the samples to reduce bandwidth.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Basic Voice Encoding: Converting Digital to Analog

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Basic Voice Encoding: Converting Digital to Analog

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Digital-to-Analog Conversion Steps 1. Decompress the samples, if compressed. 2. Decode the samples into voltage amplitudes, rebuilding the PAM signal. 3. Reconstruct the analog signal from PAM signals.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v The Nyquist Theorem

© 2006 Cisco Systems, Inc. All rights reserved.ONT v The Nyquist Theorem Sampling rate affects the quality of the digitized signal. Nyquist theorem determines the minimum sampling rate of analog signals. Nyquist theorem states that the sampling rate has to be at least twice the maximum frequency.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Example: Sampling of Voice Human speech uses 200–9,000 Hz. Human ear can sense 20–20,000 Hz. Traditional telephony systems were designed for 300–3,400 Hz. Sampling rate for digitizing voice was set to 8,000 samples per second, allowing frequencies up to 4,000 Hz.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Quantization

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Quantization Quantization is the representation of amplitudes by a certain value (step). Scale with 256 steps is used for quantization. Samples are rounded up or down to closer step. Rounding introduces inexactness (quantization noise).

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Quantization Techniques Linear quantization: –Lower signal-to-noise ratio (SNR) on small signals –Higher SNR on large signals Logarithmic quantization provides uniform SNR for all signals: –Provides higher granularity for lower signals –Corresponds to the logarithmic behavior of the human ear

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Example: Quantization of Voice There are two methods of quantization: –Mu-law, used in Canada, U.S., and Japan –A-law, used in other countries Both methods use a quasi-logarithmic scale: –Logarithmic segment sizes –Linear step sizes (within a segment) Both methods have eight positive and eight negative segments, with 16 steps per segment.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Digital Voice Encoding Each sample is encoded using eight bits: –One polarity bit –Three segment bits –Four step bits Required bandwidth for one call is 64 kbps (8000 samples per second, 8 bits each). Circuit-based telephony networks use TDM to combine multiple 64-kbps channels (DS-0) to a single physical line.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Compression Bandwidth Requirements

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Voice Codec Characteristics Standard, CodecBit Rate (kbps)Voice Quality (MOS) G.711, PCM644.1 G.726, ADPCM16, 24, (with 32 kbps) G.728, LDCELP G.729, CS-ACELP83.92 G.729A, CS-ACELP83.9

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Mean Opinion Score

© 2006 Cisco Systems, Inc. All rights reserved.ONT v What is a DSP?

© 2006 Cisco Systems, Inc. All rights reserved.ONT v What Is a DSP? A DSP is a specialized processor used for telephony applications: Voice termination: –Converts analog voice into digital format (codec) and vice versa –Provides compression, echo cancellation, VAD, CNG, jitter removal, and so on Conferencing: Mixes incoming streams from multiple parties Transcoding: Translates between voice streams that use different, incompatible codecs DSP Module Voice Network Module

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Example: DSP Used for Conferencing DSPs can be used in single- or mixed-mode conferences: –Mixed mode supports different codecs. –Single mode demands that the same codec to be used by all participants. Mixed mode has fewer conferences per DSP.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Example: DSP Used for Transcoding

© 2006 Cisco Systems, Inc. All rights reserved.ONT v Summary Whenever voice should be digitally transmitted, analog voice signals first have to be converted into digital. Conversion includes sampling, quantization, and encoding. Digitized voice has to be converted back to analog signals before being played out. Digital-to-analog conversion includes decoding and reconstruction of analog signals. The Nyquist theorem states the necessary sampling rate when converting analog signals to digital. Quantization is the process of representing the amplitude of a sampled signal by a binary number. Available codecs differ in their bandwidth requirements and voice quality. DSPs provide functions for call termination, conferences, and transcoding.

© 2006 Cisco Systems, Inc. All rights reserved.ONT v