Content Overview This module describes the concepts surrounding VoIP deployments. It also presents the advantages of converged networks and the principles of Cisco VoIP implementations. In addition, the theoretical approach to and the practical foundation for digitizing voice, IP encapsulation, and bandwidth consumption are discussed. Web Links Gateway Protocols: Waveform Coding Techniques
http://www.cisco.com/en/US/tech/tk1077/
technologies_tech_note09186a00801149b3.shtml Internetworking Multimedia, Jon Crowcroft, Mark Handley, Ian Wakeman, UCL Press
http://www.cs.ucl.ac.uk/staff/jon/mmbook/book/
index.html The Scientist and Engineer's Guide to Digital Signal Processing by Steven W. Smith, Ph.D
http://www.dspguide.com/

Content 2.1 Introducing VoIP Networks 2.1.1 Benefits of VoIP Networks Modern companies are now using converged networks, combining both data and telephony with a single IP network infrastructure. There are many benefits to using VoIP networks:

More efficient use of bandwidth and equipment: Traditional telephony networks use a 64-kbps channel for every voice call. VoIP shares bandwidth among multiple logical connections.
Lower transmission costs: A substantial amount of equipment is needed to combine 64 kbps channels into high-speed links for transport across the network. VoIP statistically multiplexes voice traffic alongside data traffic. This consolidation provides substantial savings on capital equipment and operations costs.
Consolidated network expenses: Instead of operating separate networks for voice and data, voice networks are converted to use packet-switched architecture to create a single integrated communications network with a common switching and transmission system. This results in significant cost savings over the long term on network equipment and operations.
Improved employee productivity through features provided by IP telephony: IP phones are not only telephones; they are complete business communication devices. IP phones offer directory lookups and access to databases through Extensible Markup Language (XML) applications. These applications allow simple integration of telephony into any business application. For instance, employees can use the telephone to look up information about a customer who called in, search for inventory information, and enter orders. The employee can be notified of an issue (for example, a change of the shipment date) and, with a single push of a button, can call the customer about the change. In addition, software-based telephones or wireless telephones offer mobility to the telephone user.
Access to new communication devices: Packet technology can reach devices that are largely inaccessible to the modern time-division multiplexing (TDM) infrastructures. Examples of such devices are computers, wireless devices, household appliances, personal digital assistants (PDAs), and cable set-top boxes. Intelligent access to such devices enables companies and service providers to increase the volume of communications that companies deliver, the breadth of services that companies offer, and the number of subscribers that companies serve. Packet technology, therefore, enables companies to market new devices, including video telephones, multimedia terminals, and advanced IP phones.

Although packet technology has clear benefits, businesses must consider the following points carefully before migrating to this technology:

Return on investment (ROI), when based on the new system features, can be difficult to prove.
Generally, voice and data personnel do not use the same terminology. These differences can create communication difficulties.
Current voice telephony components may not yet have fully depreciated. Replacing equipment that still has financial value may decrease the ROI of any change.

Content 2.1 Introducing VoIP Networks 2.1.2 Components of a VoIP Network A VoIP network has several components as shown in Figure . These are the most common elements in VoIP networks:

Telephones: Telephones can be IP phones, software-based telephones operated on PCs, or traditional telephones (analog or ISDN).
Gateways: Gateways interconnect the VoIP network with traditional telephony devices. Gateways are usually voice-enabled routers and provide the following functions:
- On one side, the telephone line plugs into the gateway. The gateway connects to the PSTN and communicates with any telephone in the world.
- On the other side, the gateway connects to the IP network and communicates with any computer in the world.
- The gateway takes the standard telephone signal, digitizes it (if it is not already digital), compresses it, packetizes using IP, and routes it to a destination over the IP network.
- The gateway reverses the operation for packets coming in from the network and going out to the phone.
- Both operations (coming from and going to the phone network) take place at the same time, allowing a full-duplex (two-way) conversation.
Multipoint control units: A multipoint control unit (MCU) is required for conferences. If more than two parties are involved in a call, all members of the conference send their media to the MCU. The MCU mixes the media and then sends the media to all participants.
Application servers: Application servers provide XML-based services to IP phones. IP phone users have access to directories and databases through XML applications.
Gatekeepers: Gatekeepers are useful, but optional, VoIP network components. They provide routing and central management of all endpoints (terminals, gateways, and MCUs) in a given zone. The gatekeeper and the endpoints administered from it form a management zone. Gatekeepers provide Call Admission Control (CAC) to prevent the network from being over subscribed. As well, CAC translates telephone numbers or names to IP addresses for call routing in an H.323 network.
Call agents: Call agents provide call control, CAC, bandwidth control, and address translation services to IP phones or Media Gateway Control Protocol (MGCP) gateways. Cisco CallManager is a call agent. CallManager tracks all active VoIP network components including include telephones, gateways, conference bridges, transcoding resources, and voicemail boxes among others. CallManager often uses Skinny Client Control Protocol (SCCP) for signaling the hardware endpoints of the system, such as IP phones. H.323, MGCP, or Session Initiation Protocol (SIP) is used to pass call signaling to gateways. In some ways, CallManager acts as an IP PBX.
Video endpoints: Video endpoints provide video telephony features to users. As with audio-only calls, video calls need a multipoint control unit for conferences. For videoconferences, the multipoint control unit has to be capable of mixing video and audio streams.

Figure shows how both legacy analog and VoIP components coexist in the same VoIP network. Analog and IP phones also coexist on the same network. Once companies decide to migrate to VoIP, the companies can choose an all-IP option or retain all or part of their analog networks as shown in Figure . Maintaining analog networks requires analog-to-IP conversion.

Content 2.1