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  • 5. Signaling, Protocol and Management Modules

    The VoIP software performs telephony signaling to detect the presence of a new call and to collect address (dial digit) information, which is used by the system to route a call to a destination port. It supports a wide variety of telephony-signaling protocols and can be adaptable to many environments. The software and configuration data for the voice card can be downloaded from a network-management system to allow customization, easy installation, and remote upgrades.

    The software interacts with the DSP for tone detection and generation, as well as mode of operation control based on the line supervision, and interacts with the telephony interface for signaling functions. The software receives configuration data from the network-management agent and utilizes operating-system services.

    Telephony-Signaling Gateway Module

    Figure 6 diagrams the architecture of the signaling software, which consists of the following components:


      telephony interface unit software—This periodically monitors the signaling interfaces of the module and provides basic debouncing and rotary digit collection for the interface.>
      signaling protocol unit—This contains the state machines implementing the various telephony-signaling protocols, such as E&M.


      network control unit—This maps telephony-signaling information into a format compatible with the packet voice session establishment signaling protocol.


      address translation unit—This maps the E.164 dial address to an address that can be used by the packet network (e.g., an IP address or a data link connection indentifier (DLCI) for a frame-relay network).


      DSP interface driver—This relays control information between the host microprocessor and DSPs.


      DSP downline loader—This is responsible for downline load of the DSPs at start-up, configuration update, or mode changes (e.g., switching from voice mode to fax mode when fax tones are detected).

    Figure 6. Signaling Modules

    Figure 6

    Network-Protocol Module


      IP signaling stack—This involves H.323 call control and transport software, including H.225, H.245, RTP/real-time conferencing protocol (RTCP) transport protocol, transmission control protocol (TCP), IP, and user datagram protocol (UDP).


      ATM signaling protocol stack—ATM Forum VToA voice-encapsulation protocol. ATM Forum–compliant, user-network interface (UNI) signaling protocol stack for establishing, maintaining, and clearing point-to-point and point-to-multipoint switched virtual circuits (SVCs).


      frame-relay protocol stack—This includes Frame Relay Forum VoFR voice-encapsulation protocol, permanent virtual circuit (PVC) and SVC support, local management interface (LMI), congestion management, traffic monitoring, and committed information range (CIR) enforcement.

    Network-Management Module

    The network-management software consists of three major services addressed in the MIB:


      physical interface to the telephone endpoint
      voice channel service for the following:

        processing signaling on a voice channel
        converting between PCM samples and compressed voice packets

      call-control service for parsing call-control information and establishing calls between telephony endpoints

    The VoIP software is configured and maintained through the use of a proprietary voice service MIB.

    More than 165 tutorials now available on CD-ROM for professionals on the go <%= dc_link %> Download PDF
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    RELATED PRODUCT PROFILES:
    Texas Instruments - TI VoIP Solutions
    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
    Comment on This Tutorial wherepage();   Copyright © 2004 International Engineering Consortium

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    FonoSIP.com. We provide Internet phone service with free Internet calling and unlimited US, Canada, Europe 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. FonoSIP.com 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 PBX software for businesses, resellers, ITSPs and campus applications.