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  • 4. 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:


      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 resampling of output samples to the digital interface to avoid sample slips.


      tone generator—This generates dual-tone multifrequency (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 playout unit—This buffers voice packets received from the packet network and sends them to the voice codec for playout.

    The following features are supported:


      a first in, first out (FIFO) buffer that stores voice code words before playout removes timing jitter from the incoming packet sequence
      a continuous-phase resampler that removes timing-frequency offset without causing packet slips or loss of data for voice- or voiceband-modem signals
      a timing jitter measurement that allows adaptive control of FIFO delay

    The voice-packetization protocols use a sequence-number field in the transmit packet stream to maintain temporal integrity of voice during playout. Using this approach, the transmitter inserts the contents of a free-running, modulo-16 packet counter into each transmitted packet, allowing the receiver to detect lost packets and to reproduce silence intervals during playout properly.


      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 downline 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.

    Figure 5 diagrams the architecture of the DSP software. The DSP software processes PCM samples from the telephony interface and converts them to a digital format suitable for transmission through a packet network.

    Figure 5. Voice Packet Module

    Figure 5

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    TABLE OF CONTENTS:
    Definition and Overview
    1 VoIP Applications
    2 VoIP QoS Issues
    3 VoIP–Embedded Software Architecture
    4 Voice Packet Module
    5 Signaling, Protocol and Management Modules
    6 VoIP Summary
    7 FoIP Applications
    8 PSTN Fax-Call Procedure
    9 FoIP QoS
    10 FoIP Software Architecture
    11 FoIP Summary
    Self-Test
    Correct Answers
    Glossary
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    VoIP.brujula.net. We provide Internet phone service with free Internet calling and unlimited US, Canada and World plans. We offer prepaid phone service using our voice over IP system and an analog telephone adaptor. The solutions are designed for home phone service, business phone service, call shops and cyber cafes. VoIP.brujula.net supports Xten / Counterpath SIP softphones and Internet telephony equipment such as Sipura 2000, Sipura 3000, Cisco 186, Linksys PAP2 and RT31P2. D-Link DVG-1402SL, UTstarcom F3000. We also support Asterisk PBX and offer VoIP Resellers PBX Software and business opportunities to let entrepreneurs and businesses resell voice over Internet under their own brand name.