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Technology Models
Voice over IP
Version: 2.0 
Status: Revised – 03/01/04
Contact: Roney E. Boyd, Jr.

Purpose

To provide a core infrastructure for deploying voice communication over an IP network and to support future voice connectivity requirements.

Scope

This model defines the infrastructure for Voice over an IP Network.

Applicability

This model is applicable to all colleges and campuses within the VCCS Intranet.

MODEL

The emerging Voice over IP (VoIP) is revolutionizing telecommunicates through the convergence of voice, video, fax, and data. This relatively new technology can drastically reduce long-distance costs and provide opportunities for the colleges to expand existing voice services and to add some new services, as they are needed. Voice over IP technology will enable the VCCS to implement voice services using the local college infrastructure and establish voice communications over Net.Work.Va with other locations in Virginia that have a VoIP capable infrastructure and a connection to Net.Work.Va.

This model defines the framework necessary to implement VoIP in support of the following VCCS applications:
  • In-state long distance services
  • Interactive Voice Response for the Student Information System (SIS)
  • LAN-base PBX services

    • While the technology has greatly improved over the last several years, many of the actual products required to deliver enhanced services are still emerging. Yet, even with this delay the technology is gaining wider acceptance. This model will be reviewed and updated as the technology matures and the related products are introduced into the general marketplace.

     

     

    VoIP STANDARD

    The emerging Voice over IP industry has formed a Voice over IP (VoIP) Forum within the International Multimedia Teleconferencing Consortium (IMTC) industry group. The VoIP Forum has adopted the International Telecommunications Union (ITU) H.323 suite of protocols as the basis for Voice over IP interoperability. Voice over IP uses the H.323 protocol functions that support Control (Call signaling, Channel Setup and Terminal to Gatekeeper Signaling), Real - Time Audio (Compression) and Real-Time Protocol (RTP). The other protocol functions defined within the H.323 standards are used for Desktop video. The VCCS will closely follow the work of the VoIP Forum and other recognized groups in the industry who are cooperating to develop standards for this rapidly growing industry.

     STRATEGY

    When the VCCS elected to implement Voice over the Network, using IP, it was determined that it would consist of two phases.

    Phase I

    Phase I of Voice over IP implementation would require use of the existing telecommunication equipment i.e. PBX's, Centrex Units and telephone handsets. In order to accomplish this phase, an IP Gateway device would be installed at each of the VCCS Campuses. The IP Gateway device connects to Network Virginia, the local area network (LAN) and to the traditional PBX or Centrex Unit. The implementation of the first phase, with the essential IP Gateway device, would provide the following:
  • The capability for all calls placed within the VCCS network to be a local call to any VCCS Campus.
  • Eliminate trunk tie lines between Campuses.
  • Provide voice connection to other voice application devices such as Interactive Voice Response (IVR) units.

    • Phase I Components

    Public Broadcast Exchange (PBX)

    The Public Broadcast Exchange (PBX) provides traditional voice communication between telephone handsets and the Public Switched Telephone Networks (PSTN). A PBX provides the following functions: Call Control Management, Time Slot Interchange, and Telephone Service Features. The PBX has line(s) that connect to the telephone handset, tie line(s) that connect to other PBX’s and direct trunk lines that connect to the local telephone company Central Office (CO).

    Centrex Unit

    A Centrex unit provides voice communication between telephone handsets, a Key Telephone System (KTS) or a PBX to the Public Switched Telephone Networks. Centrex units are normally located at the Local Telephone Company Central Office (CO). Connection to a Centrex unit allows the customer to use all the functions and features of a PBX without purchasing a PBX. A Centrex unit provides the same line connections as a PBX.

    IP Gateway Device

    The IP Gateway device provides signaling, connectivity and translation of traditional voice communications across an IP network to another IP Gateway device. An IP Gateway device provides the following functions:
  • Dial Plan - mapping telephone numbers to destination IP Addresses.
  • Call Setup - exchanges compatibility information, and performs any option negotiation and security handshake.
  • Digitizing - converts analog signals into packets.
  • Demodulation/Remodulation - processing signaling according to it type
  • Compression/Decompression - function by which voice data is processed to use less bandwidth.

    • An IP Gateway has line connections for traditional telephone handsets, a trunk tie line connection for other IP Gateway device. It also has a LAN interface connection and a WAN interface connection.

    Interoperability demands that the IP Gateway be compliant with International Telecommunications Union-Telecommunication Sector's (ITU-T) H.323 body of standards.

    The customer would continue to use all the functions and features that are currently incorporated in the traditional voice communication equipment i.e. voice mail, call forward, call transfer, conference calls, etc.

    Figure 1 - Phase One Implementation

    Figure 1 depicts the Phase I connectivity and hardware requirements to support In-state long distance services and Interactive Voice Response for the Student Information System.

    Phase II

    Phase II of the Voice over IP implementation would replace the traditional PBX and Centrex units with a LAN-based PBX. The LAN-based PBX provides the same functions and features as the traditional PBX or Centrex unit. Listed below are some of the advantages and benefits of moving from the traditional PBX or Centrex unit to a LAN-based PBX.
  • Does not require two different wiring structures for communication (current infrastructure requires a wire for voice and a different wire for data).
  • Does not require a proprietary Computer Telephony Integration link to bring add-on telephone functionality to the desktop computer (No additional line required from the local Telephone Company to add new users).
  • One engineer can support both the LAN and the voice service.
  • New features and options for customers are added via Graphical User Interface (GUI) software.
  • Little or no customer training requirements.
  • Reduction in overall telecommunication expense.
  • Interoperability with other H.323 compliant devices.

    • Phase II Components

    Terminal Device

    The terminal device is an IP node with a TCP/IP stack and an IP address. There are two general classes of terminal devices. The first is a desktop computer with software communications applications such as VocalTec's Internet Phone or Microsoft's NetMeeting and a sound card that performs audio encode/decode and acoustic echo cancellation. The second is an intelligent PBX phone that connects directly to an Ethernet network.

    An intelligent PBX phone, such as the Ethernet Phone, has the appearance of a traditional telephone and supports all the existing PBX services. To the customer, there is no difference in usage, but some type of gateway server is required to communicate with off network customers and LAN customers that do not have an Ethernet phone.

    Call Processor Manager

    The Call Processor Manager component contains the intelligence to enable supplementary PBX services such as call forward, call transfer and multiple line appearances. The Call Processor Manager also provides basic connectivity (call setup, signaling, etc.) and call-routing services to devices on the network, acting as a proxy among those devices. Thus, because of its call routing capability a Dial Plan is required for the Call Processor Manager. The Call Processor Manager can also function as a Gatekeeper with the registration of H.323 nodes.

    Interoperability demands that the Call Processor Manager be compliant with International Telecommunications Union-Telecommunication Sector's (ITU-T) H.323 body of standards.

    IP Gateway Device

    The IP Gateway component defined for Phase II of the Voice over IP implementation performs the same functions and has the same connections as the IP Gateway device defined in Phase I.

    The Call Processor Manager component and the IP Gateway component in Phase II perform similar functions, but for different environments. For example, an enterprise network could have multiple Call Processor Managers installed with just one IP Gateway device to provide off net access. The Call Processor Managers would provide the call setup, signaling and other service features for the IP Phones. The IP Gateway would provide call setup and signaling to a PBX or Centrex unit for off network outgoing and incoming calls for the IP Phones.

    Figure 2 - Phase Two Implementation

    Figure 2 depicts the connectivity and hardware requirements to implement LAN-based PBX services.

    CONFIGURATION REQUIREMENTS FOR PHASE I and PHASE II

    Both the IP Gateway and the Call Processor Manager require some type of dial mapping plan. Although most people are not acquainted with Dial Plans by name, they have become accustomed to using them. The Public Switch Telephone Network is designed around a 10-digit Dial Plan consisting of area code and 7-digit telephone numbers. Features within a local telephone-switching machine (such as Centrex) allow for the use of a custom 5-digit dial plan for specific customers who subscribe to that service. PBX's also allow for variable length Dial Plans containing three to 11 digits. Dial Plans contain specific dialing patterns for a customer who wants to reach a particular telephone number. If the Dial Plan is develop for a private internal voice network that is not accessed by an outside voice network, i.e. PBX or Centrex, the telephone numbers can be any number of digits, otherwise the standard 11 digit maximum should be used.

    Dial Plans require knowledge of the customer's network topology, current telephone number dialing patterns, proposed router locations and traffic routing requirements.

    IP VoIP Gateway Dial Plan Mapping consists of VoIP and Pots Statements that contain:
  • A Destination-Pattern, which equate to the telephone number and can be the full 11-digits or some mask of the telephone that compose a range of numbers.
  • A Type that identifies if the telephone number is attached to a POTS (Plain Old Telephone System) or if the telephone has to traverse the Voice over IP (VoIP) network.
  • A Voice Port that identifies which port the telephone number is connected.
  • A Session Target IP address, which identifies the IP Gateway  that outgoing telephone numbers are connected or contains the IP Address to which the telephone number relates.

    • Call Manager Dial Plan consist of Route Patterns Definitions and a Gatekeeper Definitions:

  • Route Pattern Definitions consist of a telephone number that can be in the form of digits (804XXXXXXX) or digits and symbols (9.@). It also points to a Gateway or Gatekeeper for the routing of calls to a PSTN or over a VoIP Network.  The Route Pattern is comparable to the destination, session target and port statements in an IP VoIP Gateway.

  • Gatekeeper Definitions contain a pointer to the Gatekeeper IP Address, that contains destination patterns and/or route patterns and the IP Address from which the destination pattern and/or route pattern was registered.

    • The success of any Voice over IP implementation requires that a comprehensive dial plan be developed before as part of the overall planning.

    ROLES and RESPONSIBILITIES:

     

    Technical support:

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