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