Transcript Mod 2

Optimizing
Converged Cisco
Networks (ONT)
Module 2: Cisco VoIP
Implementations
© 2006 Cisco Systems, Inc. All rights reserved.
Lesson 2.1:
Introducing
VoIP Networks
© 2006 Cisco Systems, Inc. All rights reserved.
Objectives
 Describe the benefits of a VoIP network.
 Describe the components of a VoIP network.
 Describe the legacy analog interfaces used in VoIP
networks.
 Describe the digital interfaces used in VoIP networks.
 Explain the 3 phases of call control.
 Compare and contrast distributed and centralized call
control.
© 2006 Cisco Systems, Inc. All rights reserved.
Benefits of a VoIP Network




More efficient use of bandwidth and equipment
Lower transmission costs
Consolidated network expenses
Improved employee productivity through features
provided by IP telephony:
IP phones are complete business communication devices.
Directory lookups and database applications (XML)
Integration of telephony into any
business application
Software-based and wireless
phones offer mobility.
 Access to new communications devices
(such as PDAs and cable set-top boxes)
© 2006 Cisco Systems, Inc. All rights reserved.
Components of a VoIP Network
© 2006 Cisco Systems, Inc. All rights reserved.
Legacy Analog and VoIP Applications Can
Coexist
© 2006 Cisco Systems, Inc. All rights reserved.
Legacy Analog Interfaces in VoIP Networks
Analog Interface Type
Label
Description
Foreign Exchange Station
FXS
Used by the PSTN or PBX side of an FXS–FXO connection
Foreign Exchange Office
FXO
Used by the end device side of an FXS–FXO connection
Earth and Magneto
E&M
Trunk, used between switches
© 2006 Cisco Systems, Inc. All rights reserved.
Legacy Analog Interfaces in VoIP Networks
5
1
3
1
© 2006 Cisco Systems, Inc. All rights reserved.
2
4
Digital Interfaces
Interface
Voice Channels (64 kbps Each)
Signaling
Framing
Overhead
Total
Bandwidth
BRI
2
1 channel (16 kbps)
48 kbps
192 kbps
T1 CAS
24 (no clean 64 kbps because of
robbed-bit signaling)
in-band (robbed-bits
in voice channels)
8 kbps
1544 kbps
T1 CCS
23
1 channel (64 kbps)
8 kbps
1544 kbps
E1 CAS
30
64 kbps
64 kbps
2048 kbps
E1 CCS
30
1 channel (64 kbps)
64 kbps
2048 kbps
© 2006 Cisco Systems, Inc. All rights reserved.
Call Setup
 Checks call-routing configuration
 Determines bandwidth availability
If bandwidth is available, setup message is passed
If bandwidth is not available, busy signal is generated
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Call Maintenance
 Tracks quality parameters:
Packet loss
Jitter
Delay
 Maintains or drops call based on connection quality
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Call Teardown
 Notifies devices to free resources
 Resources are made available to subsequent calls
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Distributed Call Control
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Centralized Call Control
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Self Check
1. Which type of call control uses a call agent to route
the call?
2. What is a DSP?
3. Name 3 types of analog interfaces used at gateways.
4. What are the 3 components of basic call control?
5. What phase of call control involves determining if
bandwidth is available to place the call?
© 2006 Cisco Systems, Inc. All rights reserved.
Summary
 The benefits of a VoIP network include more efficient
use of network bandwidth and equipment, lower cost
and consolidated expenses.
 Legacy analog and VoIP applications and devices can
coexist.
 The 3 stages of a VoIP call include call setup, call
maintenance, and call teardown.
 VoIP can be deployed in a centralized or distributed
environment.
© 2006 Cisco Systems, Inc. All rights reserved.
Lesson 2.2:
Digitizing
and
Packetizing
Voice
© 2006 Cisco Systems, Inc. All rights reserved.
Objectives
 Describe the process of analog to digital conversion.
 Describe the process of digital to analog conversion.
 Explain how sampling rates are determined using the
Nyquist Theorem.
 Explain how quantization can lead to noise.
 Explain how MOS is used to judge voice quality.
 Describe the purpose of DSPs.
© 2006 Cisco Systems, Inc. All rights reserved.
Basic Voice Encoding: Converting Analog
Signals to Digital Signals
 Step 1: Sample the analog signal.
 Step 2: Quantize sample into a binary expression.
 Step 3: Compress the samples to reduce bandwidth.
© 2006 Cisco Systems, Inc. All rights reserved.
Basic Voice Encoding:
Converting Digital Signals to Analog Signals
 Step 1: Decompress the samples.
 Step 2: Decode the samples into voltage amplitudes, rebuilding
the PAM signal.
 Step 3: Reconstruct the analog signal from the PAM signals.
© 2006 Cisco Systems, Inc. All rights reserved.
Determining Sampling Rate with the Nyquist
Theorem
 The sampling rate affects the quality of the digitized signal.
 Applying the Nyquist theorem determines the minimum sampling
rate of analog signals.
 Nyquist theorem requires that the sampling rate has to be at least
twice the maximum frequency.
© 2006 Cisco Systems, Inc. All rights reserved.
Example: Setting the Correct Voice Sampling
Rate
 Human speech uses 200–9000 Hz.
 Human ear can sense 20–20,000 Hz.
 Traditional telephony systems were designed for
300–3400 Hz.
 Sampling rate for digitizing voice was set to 8000
samples per second, allowing frequencies up to 4000
Hz.
© 2006 Cisco Systems, Inc. All rights reserved.
Quantization
 Quantization is the representation of amplitudes by a
certain value (step).
 A scale with 256 steps is used for quantization.
 Samples are rounded up or down to the closer step.
 Rounding introduces inexactness (quantization noise).
© 2006 Cisco Systems, Inc. All rights reserved.
Quantization Techniques
 Linear quantization:
Lower SNR on small signals (worse voice quality)
Higher SNR on large signals (better voice quality)
 Logarithmic quantization provides uniform SNR for all
signals:
Provides higher granularity for lower signals
Corresponds to the logarithmic behavior of the human ear
© 2006 Cisco Systems, Inc. All rights reserved.
Digital Voice Encoding
 Each sample is encoded using eight bits:
One polarity bit
Three segment bits
Four step bits
 Required bandwidth for one call is 64 kbps
(8000 samples per second, 8 bits each).
 Circuit-based telephony networks use TDM to combine
multiple 64-kbps channels (DS-0) to a single physical
line.
© 2006 Cisco Systems, Inc. All rights reserved.
Companding
 Companding — compressing and expanding
 There are two methods of companding:
Mu-law, used in Canada, U.S., and Japan
A-law, used in other countries
 Both methods use a quasi-logarithmic scale:
Logarithmic segment sizes
Linear step sizes (within a segment)
 Both methods have eight positive and eight negative
segments, with 16 steps per segment.
 An international connection needs to use A-law; mu-toA conversion is the responsibility of the mu-law country.
© 2006 Cisco Systems, Inc. All rights reserved.
Coding
 Pulse Code Modulation (PCM)
Digital representation of analog signal
Signal is sampled regularly at uniform levels
Basic PCM samples voice 8000 times per second
Basis for the entire telephone system digital hierarchy
 Adaptive Differential Pulse Code Modulation
Replaces PCM
Transmits only the difference between one sample and the next
© 2006 Cisco Systems, Inc. All rights reserved.
Common Voice Codec Characteristics
ITU-T
Standard
Codec
Bit Rate (kbps)
G.711
PCM
G.726
ADPCM
G.728
LDCELP (Low Delay CELP)
16
G.729
CS-ACELP
8
G.729A
CS-ACELP, but with less
computation
8
© 2006 Cisco Systems, Inc. All rights reserved.
64
16, 24, 32
Mean Opinion Score
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A Closer Look at a DSP
A DSP is a specialized processor
used for telephony applications:
DSP Module
 Voice termination:
Works as a compander converting
analog voice to digital format and
back again
Provides echo cancellation, VAD,
CNG, jitter removal, and other
benefits
 Conferencing: Mixes incoming
streams from multiple parties
 Transcoding: Translates between
voice streams that use different,
incompatible codecs
© 2006 Cisco Systems, Inc. All rights reserved.
Voice Network Module
DSP Used for Conferencing
 DSPs can be used in
single- or mixed-mode
conferences:
Mixed mode supports
different codecs.
Single mode demands that
the same codec to be used
by all participants.
 Mixed mode has fewer
conferences per DSP.
© 2006 Cisco Systems, Inc. All rights reserved.
Example: DSP Used for Transcoding
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Self Check
1. What sampling frequency is recommended by the
Nyquist Theorem for reconstruction of a signal?
2. What is the Hz range for traditional telephone
systems?
3. What is the implication of using 8 bits for
quantization?
4. What is the purpose of logarithmic quantization?
5. What is MOS?
© 2006 Cisco Systems, Inc. All rights reserved.
Summary
 Voice-enabled routers convert analog voice signals to
digital format for encapsulation in IP packets and
transport over IP networks. These packets are
converted back to analog at the other end.
 Quantization is the process of selecting binary values to
represent voltage levels of voice samples. Quantization
errors arise when too few samples are taken.
 There are two methods of companding: Mu-law, used in
Canada, U.S., and Japan, and A-law, used in other
countries.
 The Mean Opinion Score (MOS) provides a numerical
indication of the perceived quality of received media
after compression and/or transmission.
© 2006 Cisco Systems, Inc. All rights reserved.
Q and A
© 2006 Cisco Systems, Inc. All rights reserved.
Resources
 Voice Codec Bandwidth Calculator (requires CCO
login)
http://tools.cisco.com/Support/VBC/do/CodecCalc1.do
 DSP Calculator
http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl
 Free VoIP Quality Tester
http://www.testyourvoip.com/
© 2006 Cisco Systems, Inc. All rights reserved.
Lesson 2.3:
Encapsulating
Voice Packets
for Transport
© 2006 Cisco Systems, Inc. All rights reserved.
Objectives
 Compare and contrast voice transport in circuitswitched and VoIP networks.
 Describe causes of and solutions for jitter.
 Explain the issues with IP, TCP, and UDP when
transporting voice packets.
 Describe encapsulation overhead issues for VoIP.
 Describe header compression and when and where it
should be used.
© 2006 Cisco Systems, Inc. All rights reserved.
Voice Transport in Circuit-Switched Networks
 Analog phones connect to CO switches.
 CO switches convert between analog and digital.
 After call is set up, PSTN provides:
End-to-end dedicated circuit for this call (DS-0)
Synchronous transmission with fixed bandwidth and very low, constant delay
© 2006 Cisco Systems, Inc. All rights reserved.
Voice Transport in VoIP Networks
 Analog phones connect to voice gateways.
 Voice gateways convert between analog and digital.
 After call is set up, IP network provides:
Packet-by-packet delivery through the network
Shared bandwidth, higher and variable delays
© 2006 Cisco Systems, Inc. All rights reserved.
Jitter
 Voice packets enter the network at a constant rate.
 Voice packets may arrive at the destination at a
different rate or in the wrong order.
 Jitter occurs when packets arrive at varying rates.
 Since voice is dependent on timing and order, a
process must exist so that delays and queuing issues
can be fixed at the receiving end.
 The receiving router must:
Ensure steady delivery (delay)
Ensure that the packets are in the right order
© 2006 Cisco Systems, Inc. All rights reserved.
VoIP Protocol Issues
 IP does not guarantee reliability, flow control, error
detection or error correction.
 IP can use the help of transport layer protocols TCP or
UDP.
 TCP offers reliability, but voice doesn’t need it…do not
retransmit lost voice packets.
 TCP overhead for reliability consumes bandwidth.
 UDP does not offer reliability. But it also doesn’t offer
sequencing…voice packets need to be in the right
order.
 RTP, which is built on UDP, offers all of the functionality
required by voice packets.
© 2006 Cisco Systems, Inc. All rights reserved.
Protocols Used for VoIP
Feature
Voice
Needs
TCP
Reliability
No
Yes
Reordering
Yes
Timestamping
Yes
No
Overhead
As little as
possible
Contains
unnecessary
information
Multiplexing
Yes
© 2006 Cisco Systems, Inc. All rights reserved.
Yes
Yes


UDP
No

RTP
No

No
Yes

No
Yes

Low
Yes


Low

No
Voice Encapsulation
 Digitized voice is encapsulated into RTP, UDP, and IP.
 By default, 20 ms of voice is packetized into a single IP
packet.
© 2006 Cisco Systems, Inc. All rights reserved.
Voice Encapsulation Overhead
 Voice is sent in small packets at high packet rates.
 IP, UDP, and RTP header overheads are enormous:
For G.729, the headers are twice the size of the payload.
For G.711, the headers are one-quarter the size of the payload.
 Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring
Layer 2 overhead.
© 2006 Cisco Systems, Inc. All rights reserved.
RTP Header Compression
 Compresses the IP, UDP, and RTP headers
 Is configured on a link-by-link basis
 Reduces the size of the headers substantially
(from 40 bytes to 2 or 4 bytes):
4 bytes if the UDP checksum is preserved
2 bytes if the UDP checksum is not sent
 Saves a considerable amount of bandwidth
© 2006 Cisco Systems, Inc. All rights reserved.
cRTP Operation
Condition
Action
The change is
predictable.
The sending side tracks the predicted
change.
The predicted change The sending side sends a hash of the
is tracked.
header.
The receiving side
predicts what the
constant change is.
The receiving side substitutes the original
stored header and calculates the
changed fields.
There is an
unexpected change.
The sending side sends the entire header
without compression.
© 2006 Cisco Systems, Inc. All rights reserved.
When to Use RTP Header Compression
 Use cRTP:
Only on slow links (less than 2 Mbps)
If bandwidth needs to be conserved
 Consider the disadvantages of cRTP:
Adds to processing overhead
Introduces additional delays
 Tune cRTP—set the number of sessions to be
compressed (default is 16).
© 2006 Cisco Systems, Inc. All rights reserved.
Self Check
1. What causes jitter?
2. Explain why IP is not well suited to voice
transmission.
3. What issues does TCP have when considering it as
the protocol for voice?
4. What guidelines can be used to determine when and
where to use RTP header compression?
5. What are some disadvantages to using RTP header
compression?
© 2006 Cisco Systems, Inc. All rights reserved.
Summary
 Circuit-switched calls use dedicated links. VoIP
networks send voice in packets.
 Jitter is caused by packets arriving at the destination at
varying rates and not in the original order.
 IP, TCP or UDP alone cannot be used for voice
packets. IP does not guarantee reliability, flow control,
error detection or error correction. TCP has
unnecessary overhead. UDP needs additional
functionality offered by RTP.
 Encapsulation overhead for VoIP can be very large.
Since voice packets are small, compression should be
used to compress headers.
© 2006 Cisco Systems, Inc. All rights reserved.
Q and A
© 2006 Cisco Systems, Inc. All rights reserved.
Resources
 Examples of Jitter impact on quality
http://www.voiptroubleshooter.com/problems/jitter.html
 Understanding Jitter in Packet Voice Networks (Cisco
IOS Platforms)
http://cisco.com/en/US/tech/tk652/tk698/technologies_tech_note
09186a00800945df.shtml
© 2006 Cisco Systems, Inc. All rights reserved.
Lesson 2.4:
Calculating
Bandwidth
Requirements
for VoIP
© 2006 Cisco Systems, Inc. All rights reserved.
Objectives
 Describe factors influencing encapsulation overhead
and bandwidth requirements for VoIP.
 Explain how the packetization period impacts VoIP
packet size and rate.
 Explain how link encapsulation effects data-link
overhead on a per link basis.
 Explain the bandwidth impact of adding a tunneling
protocol header to voice packets.
 Use the bandwidth calculation process to calculate
bandwidth needs for various VoIP call types.
 Describe how VAD is used in VoIP implementations.
© 2006 Cisco Systems, Inc. All rights reserved.
Factors Influencing Encapsulation Overhead
and Bandwidth
Factor
Description
Packet rate
– Derived from packetization period (the
period over which encoded voice bits are
collected for encapsulation)
Packetization size
(payload size)
– Depends on packetization period
IP overhead
(including UDP and RTP)
– Depends on the use of cRTP
Data-link overhead
– Depends on protocol
(different per link)
Tunneling overhead (if
used)
– Depends on protocol (IPsec, GRE, or
MPLS)
© 2006 Cisco Systems, Inc. All rights reserved.
– Depends on codec bandwidth
(bits per sample)
Bandwidth Implications of Codecs
 Codec bandwidth is for voice
information only.
Codec
Bandwidth
 No packetization overhead is
included.
G.711
64 kbps
G.726 r32
32 kbps
G.726 r24
24 kbps
G.726 r16
16 kbps
G.728
16 kbps
G.729
8 kbps
© 2006 Cisco Systems, Inc. All rights reserved.
How the Packetization Period Impacts VoIP
Packet Size and Rate
 High packetization period results in:
Larger IP packet size (adding to the payload)
Lower packet rate (reducing the IP overhead)
© 2006 Cisco Systems, Inc. All rights reserved.
VoIP Packet Size and Packet Rate Examples
Codec and
Packetization Period
G.711
20 ms
G.711
30 ms
G.729
20 ms
G.729
40 ms
Codec bandwidth
(kbps)
64
64
8
8
Packetization size
(bytes)
160
240
20
40
IP overhead
(bytes)
40
40
40
40
VoIP packet size
(bytes)
200
280
60
80
Packet rate
(pps)
50
33.33
50
25
© 2006 Cisco Systems, Inc. All rights reserved.
Data-Link Overhead Is Different per Link
Data-Link
Protocol
Ethernet
Frame
Relay
MLP
Ethernet Trunk
(802.1Q)
Overhead
[bytes]
18
6
6
22
© 2006 Cisco Systems, Inc. All rights reserved.
Security and Tunneling Overhead
 IP packets can be secured by IPsec.
 Additionally, IP packets or data-link frames can be
tunneled over a variety of protocols.
 Characteristics of IPsec and tunneling protocols are:
The original frame or packet is encapsulated into another
protocol.
The added headers result in larger packets and higher
bandwidth requirements.
The extra bandwidth can be extremely critical for voice packets
because of the transmission of small packets at a
high rate.
© 2006 Cisco Systems, Inc. All rights reserved.
Extra Headers in Security and Tunneling
Protocols
Protocol
Header Size (bytes)
IPsec transport mode
30–53
IPsec tunnel mode
50–73
L2TP/GRE
24
MPLS
4
PPPoE
8
© 2006 Cisco Systems, Inc. All rights reserved.
Example: VoIP over IPsec VPN
 G.729 codec (8 kbps)
 20-ms packetization period
 No cRTP
 IPsec ESP with 3DES and SHA-1, tunnel mode
© 2006 Cisco Systems, Inc. All rights reserved.
Total Bandwidth Required for a VoIP Call
 Total bandwidth of a VoIP call, as seen on the link, is important for:
Designing the capacity of the physical link
Deploying Call Admission Control (CAC)
Deploying QoS
© 2006 Cisco Systems, Inc. All rights reserved.
Total Bandwidth Calculation Procedure
 Gather required packetization information:
Packetization period (default is 20 ms) or size
Codec bandwidth
 Gather required information about the link:
cRTP enabled
Type of data-link protocol
IPsec or any tunneling protocols used
 Calculate the packetization size or period.
 Sum up packetization size and all headers and trailers.
 Calculate the packet rate.
 Calculate the total bandwidth.
© 2006 Cisco Systems, Inc. All rights reserved.
Bandwidth Calculation Example
© 2006 Cisco Systems, Inc. All rights reserved.
Quick Bandwidth Calculation
Total packet size
—————————
Total bandwidth requirement
=
Payload size
————————————————
Nominal bandwidth requirement
Total packet size = All headers + payload
Parameter
Value
Layer 2 header
6 to 18 bytes
IP + UDP + RTP headers
40 bytes
Payload size (20-ms sample interval)
20 bytes for G.729, 160 bytes for G.711
Nominal bandwidth
8 kbps for G.729, 64 kbps for G.711
Example: G.729 with Frame Relay:
Total bandwidth requirement =
(6 + 40 + 20 bytes) * 8 kbps
—————————————
20 bytes
© 2006 Cisco Systems, Inc. All rights reserved.
= 26.4 kbps
VAD Characteristics
 Detects silence (speech pauses)
 Suppresses transmission of “silence patterns”
 Depends on multiple factors:
Type of audio (for example, speech or MoH)
Level of background noise
Other factors (for example, language, character of speaker, or
type of call)
 Can save up to 35 percent of bandwidth
© 2006 Cisco Systems, Inc. All rights reserved.
VAD Bandwidth-Reduction Examples
Data-Link
Overhead
Ethernet
Frame Relay Frame Relay MLPP
18 bytes
6 bytes
6 bytes
6 bytes
IP overhead
no cRTP
cRTP
no cRTP
cRTP
40 bytes
4 bytes
40 bytes
2 bytes
G.711
G.711
G.729
G.729
64 kbps
64 kbps
8 kbps
8 kbps
20 ms
30 ms
20 ms
40 ms
160 bytes
240 bytes
20 bytes
40 bytes
Bandwidth
without VAD
87.2 kbps
66.67 kbps
26.4 kbps
9.6 kbps
Bandwidth with
VAD (35%
reduction)
56.68 kbps
43.33 kbps
17.16 kbps
6.24 kbps
Codec
Packetization
© 2006 Cisco Systems, Inc. All rights reserved.
Self Check
1. Describe the relationship between packetization
period and packet size and packet rate.
2. How does the data-link protocol used effect
bandwidth considerations?
3. What is the default packetization period on Cisco
devices?
4. What is VAD?
5. How much bandwidth can be saved, on average,
using VAD?
© 2006 Cisco Systems, Inc. All rights reserved.
Summary
 VoIP packet size and rate are determined by the
packetization period.
 Data-link overhead must be considered with calculating
bandwidth requirements. Different links have different
overhead requirements.
 Adding a tunneling protocol header effects the
bandwidth requirements for voice packets. This
additional overhead must be considered when
calculating bandwidth requirements.
 Voice Activity Detection (VAD) is a process used to
detect silence in order to save bandwidth. VAD can
save 34% on average.
© 2006 Cisco Systems, Inc. All rights reserved.
Q and A
© 2006 Cisco Systems, Inc. All rights reserved.
Resources
 Voice Over IP - Per Call Bandwidth Consumption
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech
_note09186a0080094ae2.shtml#topic1
 Voice Codec Bandwidth Calculator
http://tools.cisco.com/Support/VBC/do/CodecCalc1.do
© 2006 Cisco Systems, Inc. All rights reserved.
Lesson 2.5:
Implementing
VoIP in an
Enterprise
Network
© 2006 Cisco Systems, Inc. All rights reserved.
Objectives
 List the common components of an enterprise voice
implementation.
 Describe Call Admission Control and how it differs from
QoS.
 Describe the functions of the Cisco Unified
CallManager.
 Identify common enterprise IP telephony deployment
models.
 Identify basic Cisco IOS VoIP configuration commands.
© 2006 Cisco Systems, Inc. All rights reserved.
Enterprise Voice Implementations
 Components of enterprise voice networks:
Gateways and gatekeepers
Cisco Unified CallManager and IP phones
© 2006 Cisco Systems, Inc. All rights reserved.
Deploying CAC
 CAC artificially limits the number of concurrent voice calls.
 CAC prevents oversubscription of WAN resources caused by too much voice traffic.
 CAC is needed because QoS cannot solve the problem of voice call
oversubscription:
QoS gives priority only to certain packet types (RTP versus data).
QoS cannot block the setup of too many voice calls.
Too much voice traffic results in delayed voice packets.
© 2006 Cisco Systems, Inc. All rights reserved.
Example: CAC Deployment
 IP network (WAN) is only designed for two concurrent voice calls.
 If CAC is not deployed, a third call can be set up, causing poor
quality for all calls.
 When CAC is deployed, the third call is blocked.
© 2006 Cisco Systems, Inc. All rights reserved.
Voice Gateway Functions on a Cisco Router
 Connects traditional telephony devices to VoIP
 Converts analog signals to digital format
 Encapsulates voice into IP packets
 Performs voice compression
 Provides DSP resources for conferencing and
transcoding
 Supports fallback scenarios for IP phones (Cisco
SRST)
 Acts as a call agent for IP phones (Cisco Unified
CallManager Express)
 Provides DTMF relay and fax and modem support
© 2006 Cisco Systems, Inc. All rights reserved.
Cisco Unified CallManager Functions
Call processing
Dial plan administration
Signaling and device control
Phone feature administration
Directory and XML services
Programming interface to external applications
© 2006 Cisco Systems, Inc. All rights reserved.
Cisco IP Communicator
Example: Signaling and Call Processing
© 2006 Cisco Systems, Inc. All rights reserved.
Enterprise IP Telephony Deployment Models
Deployment Model
Single site
Characteristics
– Cisco Unified CallManager cluster at the single site
– Local IP phones only
Multisite with centralized
call processing
– Cisco Unified CallManager cluster only at a single
site
– Local and remote IP phones
Multisite with distributed call
processing
– Cisco Unified CallManager clusters at multiple sites
Clustering over WAN
– Single Cisco Unified CallManager cluster distributed
over multiple sites
– Local IP phones only
– Usually local IP phones only
– Requirement: Round-trip delay between any pair of
servers not to exceed 40 ms
© 2006 Cisco Systems, Inc. All rights reserved.
Single Site
 Cisco Unified CallManager
servers, applications, and DSP
resources are located at the
same physical location.
 IP WAN is not used for voice.
 PSTN is used for all external
calls.
Note: Cisco Unified CallManager
cluster can be connected to
various places depending on the
topology.
© 2006 Cisco Systems, Inc. All rights reserved.
Multisite with Centralized Call Processing
 Cisco Unified CallManager servers and applications are located at the central site
while DSP resources are distributed.
 IP WAN carries data and voice (signaling for all calls, media only for intersite calls).
 PSTN access is provided at all sites.
 CAC is used to limit the number of VoIP calls, and AAR is used if WAN bandwidth is
exceeded.
 Cisco SRST is located at the remote branch.
Note: Cisco Unified CallManager cluster can be connected to various places depending on
the topology.
© 2006 Cisco Systems, Inc. All rights reserved.
Multisite with Distributed Call Processing
 Cisco Unified CallManager servers, applications, and DSP resources are located at
each site.
 IP WAN carries data and voice for intersite calls only (signaling and media).
 PSTN access is provided at all sites; rerouting to PSTN is configured if IP WAN is
down.
 CAC is used to limit the number of VoIP calls, and AAR is used if WAN bandwidth is
exceeded.
Note: Cisco Unified CallManager cluster can be connected to various places, depending on
the topology.
© 2006 Cisco Systems, Inc. All rights reserved.
Clustering over WAN
 Cisco Unified CallManager servers of a single cluster are distributed among multiple sites while
applications and DSP resources are located at each site.
 Intracluster communication (such as database synchronization) is performed over
the WAN.
 IP WAN carries data and voice for intersite calls only (signaling and media).
 PSTN access is provided at all sites; rerouting to PSTN is performed if IP WAN is down.
 CAC is used to limit the number of VoIP calls; AAR is used if WAN bandwidth is exceeded.
Note: Cisco Unified CallManager cluster can be connected to various places, depending on the topology.
© 2006 Cisco Systems, Inc. All rights reserved.
Basic Cisco IOS VoIP Voice Commands
© 2006 Cisco Systems, Inc. All rights reserved.
Voice-Specific Commands
router(config)#
dial-peer voice tag type
 Use the dial-peer voice command to enter the dial peer subconfiguration mode.
router(config-dial-peer)#
destination-pattern telephone_number
 The destination-pattern command, entered in dial peer subconfiguration mode,
defines the telephone number that applies to the dial peer.
© 2006 Cisco Systems, Inc. All rights reserved.
Voice-Specific Commands (Cont.)
router(config-dial-peer)#
port port-number
 The port command, entered in POTS dial peer subconfiguration mode, defines the
port number that applies to the dial peer. Calls that are routed using this dial peer are
sent to the specified port.
router(config-dial-peer)#
session target ipv4:ip-address
 The session target command, entered in VoIP dial peer subconfiguration mode,
defines the IP address of the target VoIP device that applies to the dial peer.
© 2006 Cisco Systems, Inc. All rights reserved.
Self Check
1. What is CAC?
2. What can happen is CAC is not used?
3. What command is used to define the telephone
number that applies to the dial peer?
4. List 4 deployment options when using the Cisco
Unified CallManager.
© 2006 Cisco Systems, Inc. All rights reserved.
Summary
 Enterprise voice implementations use components
such as gateways, gatekeepers, Cisco Unified
CallManager, and IP phones.
 Call Admission Control (CAC) extends the functionality
of QoS to ensure that an additional call is not allowed
unless bandwidth is available to support it.
 Enterprise IP Telephony deployment models include
single site, multisite with centralized call processing,
multisite with distributed call processing, and clustering
over the WAN.
© 2006 Cisco Systems, Inc. All rights reserved.
Q and A
© 2006 Cisco Systems, Inc. All rights reserved.
Resources
 Video: The ABCs of VoIP (16 min.)
http://tools.cisco.com/cmn/jsp/index.jsp?id=43596
 Voice and Unified Communications
http://www.cisco.com/en/US/products/sw/voicesw/index.html
 VoIP Call Admission Control
http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/
cac.htm
© 2006 Cisco Systems, Inc. All rights reserved.