Voice Packet Module
This section describes the functions performed by the software in the voice packet module, also known as the voice-processing module, which is primarily responsible for processing the voice data. This function is usually performed in a DSP. The voice-processing module consists of the following software:
This section describes the functions performed by the software in the voice packet module, also known as the voice-processing module, which is primarily responsible for processing the voice data. This function is usually performed in a DSP. The voice-processing module consists of the following software:
- PCM Interface: This receives pulse code modulation (PCM) samples from the digital interface and forwards them to appropriate DSP software modules for processing, forwards processed PCM samples received from various DSP software modules to the digital interface, and performs continuous phase re-sampling of output samples to the digital interface to avoid sample slips.
- Tone Generator: This generates dual-tone multi-frequency (DTMF) tones and call progress tones under command of the host (e.g., telephone, fax, modem, PBX, or telephone switch) and is configurable for support of U.S. and international tones.
- Echo Canceller: This performs G.165–compliant echo cancellation on sampled, full-duplex voice port signals. It has a programmable range of tail lengths. voice activation detector/idle noise measurement —This monitors the received signal for voice activity. When no activity is detected for the configured period of time, the software informs the packet voice protocol. This prevents the encoder output from being transported across the network when there is silence, resulting in additional bandwidth savings. This software also measures the idle noise characteristics of the telephony interface. It reports this information to the packet voice protocol to relay this information to the remote end for noise generation when no voice is present
- Tone Detector: This detects the reception of DTMF tones and performs voice/fax discrimination. Detected tones are reported to the host so that the appropriate speech or fax functions are activated.
- Voice Codec Software: This compresses the voice data for transmission over the packet data. It is capable of numerous compression ratios through the modular architecture. A compression ratio of 8:1 is achievable with the G.729 voice codec (thus, the normal 64–kbps PCM signal is transmitted using only 8 kbps).
- Fax Software: This performs a fax-relay function by demodulating PCM data, extracting the relevant information, and packing the fax-line scan data into frames for transmission over the packet network. Significant bandwidth savings can be achieved by this process.
- Voice Play-out Unit: This buffers voice packets received from the packet network and sends them to the voice codec for play-out.
- A first in, first out (FIFO) buffer that stores voice code words before play-out removes timing jitter from the incoming packet sequence.
- A continuous-phase re-sampler that removes timing-frequency offset without causing packet slips or loss of data for voice- or voice-band-modem signals.
- A timing jitter measurement that allows adaptive control of FIFO delay.
- Packet Voice Protocol: This encapsulates compressed voice and fax data for end-to-end transmission over a backbone network between two ports.
- Control Interface Software: This coordinates the exchange of monitor and control information between the DSP and host via a mailbox mechanism. Information exchanged includes software down-line load, configuration data, and status reporting.
- Real-time Portability Environment: This provides the operating environment for the software residing on the DSP. It provides synchronization functions, task management, memory management, and timer management.
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| Figure 5: Voice Packet Module |