smben20s05l01
TRANSCRIPT
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Voice Technologies
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Module Overview
This module gives an overview of the standard telephonytechnologies and introduces the Cisco IP telephony andthe Cisco Unified Communications technology.
Upon completing this module, you will be able to describe
basic telephony concepts, including IP telephony andUnified Communications. This ability includes being ableto meet these objectives:
Explain traditional telephony basics and Cisco VoIP functionality,and apply telephony concepts to trunking options
Define infrastructure and components for IP telephony and CiscoUnified Communications
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Voice Technologies
Telephony BasicsOverview ofTelephony
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Lesson Overview
Upon completing this lesson, you will be able to:
Describe the components of traditional telephony network
Explain the primary differences between a PBX and aKey System
Explain common telephony features
List and describe the terms used to describe a telephonyconnection
Describe the functionality of gateways and their role ofconnecting IP to traditional PSTN and telephony equipment
Explain the primary differences between analog and digitaltrunking options
Describe the major VoIP signaling protocols, H.323, SIP,and SCCP
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Traditional Telephony Overview
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Important Telephony Terms
DTMF dual tone multifrequency
ITU International Telecommunications Union
ETSI European Telecommunications Standards Institute
Local Loop
Trunk
Tie Line Trunk
Central Office = CO (Local Exchange)
POTS plain old telephone service
PCM pulse code modulation
TDM time-division multiplexing
codec coder and decoder
DSP digital signal processor
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Basic Components of a TraditionalTelephony Network
Outsidecalls
In-housecalls
BostonSan Jose
EdgeDevices
Exchange Exchange
Tie Trunk
ExchangeTrunks
LocalLoop
LocalLoop
Switch Switch
PBX PBX
Tie Trunk
ExchangeTrunks
PSTN
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Pulse Code Modulation
Initial signal is analog
Analog signal converted to digital, 8 bit word format
0 0 1 1 0 0 0 1
Band Limiter,Sampler,
Quantization
Analog Signal Digital Signal
8 bit word 8 bit word.
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Time-Division Multiplexing
Provides a mechanism for transporting multiple callsbetween two points on a common carrier
Used by CAS and ISDN trunks in customer access networks andforms the basis of service-provider networks
Conversation 1
Conversation 2
Conversation 3
Conversation 4
Conversation 24/30
11 11 1
3 3 3 3 3
4 4444
24 24 2424
2 2 2 2 24 3 12 24 4 3 12
Carrier Channel
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PBX Functionality
PBX
Line Trunk LocalExchangeTerminal interface Terminal interface
Switching network Terminal interface
Control complex Terminal interface
Power supply andfans Terminal interface
LineCard
Trunkcard
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Key System Functionality
Keysystem
TrunksTerminationblocks
RJ-11
LocalExchange
CO line cards
Station cards
Station cards
Intercom cardsPower supply
Main distribution frame
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Comparing PBXs with Key Systems
Comparing Key Systems with PBXs
PBX Key System
Technology Primarily digital Analog or digital
Switch functionality Similar to the CO switch Not a switch
Typical installationLarge company site(typically more than 50 users)
Small company or branchoffice(typically 50 users or fewer)
Method for AccessingOutside Trunks Dial 9 or other access numberto access outside line Press a button to accessoutside line
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Common PBX Features
Shared lines or shared DN
Hunt groups
Monitor lines
Intercom lines
Audio paging
Speed dial
Overlaid lines
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Introducing Voice over IP
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Using IP as a Transport Medium forTelephony
IP networks are better suited tomultimedia networks as they can usebandwidth more efficiently than TDMnetworks.
A voice gateway is used to interfaceTDM PBXs and the PSTN with IPnetworks.
Voice gateways are often used to:
Transfer wide-area TDM traffic toan IP network.
Connect IP PBX systems to thePSTN or analog endpoints.
Gateways can support analog ordigital telephony interfaces.
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Analog Voice Ports
FXS: Connects directly to end-user equipment such as telephones, faxmachines, or modems
FXO: Used for trunk, or tie line, connections to a PSTN CO or to a PBX thatdoes not support E&M signaling
E&M: Used for private PBX-to-PBX connectivity
FXS
PSTNFXOFXO
E&M E&MWAN orPSTN
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Digital Voice Ports
T1-CAS: Supports 24 voice channels. Signaling is embedded inthe voice channel.
T1-PRI: Supports 23 voice (B) channels. Signaling is carried in adedicated 64kbps data (D) channel.
E1-CAS: Supports 30 voice channels. Signaling is transported ina dedicated signaling channel.
E1-PRI: Supports 30 voice (B) channels. Signaling is carried in adedicated 64kbps data (D) channel.
Basic Rate ISDN (BRI): Supports 2 voice (B) channels. Signaling
is carried in a dedicated 16kbps data (D) channel.
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Gateway Features
Gateways support multiple IP protocols such as:
H.323
SIP
SCCP
Gateways also provide additional functionality
DTMF relay
Supplementary services
Provide fax and modem services, or both
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Signaling Protocols
Protocol Description
H.323ITU standard protocol for interactive conferencing; evolved fromH.320 ISDN standard; flexible, complex
SIPIETF protocol for interactive and non-interactive conferencing; simpler,but less mature, than H.323
SCCPCisco protocol used by Unified Communications Manager and UnifiedCommunications Manager Express to control telephony endpoints.
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H.323
H.323 suite:
Approved in 1996 by the ITU-T
Peer-to-peer protocol where end devices initiate sessions
Widely used with gateways, gatekeepers, or third-party H.323
clients, especially video terminals in Cisco UnifiedCommunications
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Session Initiation Protocol (SIP)
SIP Attributes:
IETF RFC 2543 (1999), RFC 3261 (2002)
Based on the logic of the World Wide Web
Widely used with gateways and proxy servers within service
provider networks Peer-to-peer protocol where end devices (user agents) initiate
sessions
ASCII text-based for easy implementation and debugging
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How SIP Works
ASCII-based protocol.
User identified by a unique SIP address.
Users optionally register with a registrar server using their
assigned SIP addresses. When a user initiates a call, a SIP request is sent to a SIP server.
Location of end user can be dynamically registered with the SIPserver.
http://sip:[email protected]/http://sip:[email protected]/ -
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Advantages of SIP Gateways
Dial plan configuration directly on the gateway
Translations defined per gateway
Advanced support for third-party telephony system integration
Interoperability with third-party voice gateways
Support of third-party end devices (SIP phones)
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Skinny Call Control Protocol (SCCP)
Cisco proprietary lightweight client protocol used by:
Cisco Unified Communications Manager
Cisco Unified Communications Manager Express
Client messaging set controls:
Registration
Management
Call control
Media control
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Q&A
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Lesson Summary
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Lesson Summary
An underlying knowledge of the basic terminology, components,and features of traditional telephony is necessary for engineers.
Small organizations and branch offices often use a key telephonesystem because a PBX offers functionality and extra features thatmay not be required.
A voice gateway is used to interface TDM PBXs and the PSTNwith PT networks.
Cisco analog station gateways connect an IP telephony networkto POTS, whereas Cisco access digital trunk gateways connectan IP telephony network to the PSTN via digital trunks.
Cisco voice gateways support various signaling protocols,depending on the hardware platform.
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