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NETMASTERCLASS
LEARNiT: Cisco Voice over IP

Reference Notes

Program FOR Network Professionals

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LEARNiT: Cisco Voice over IP

Many complex technologies must be mastered in order to succeed with Cisco Voice over IP. This training
package allows you to master these key Cisco security technologies in the most efficient manner possible.

The LEARNiT: Cisco Voice over IP (CVOICE) package includes the following:

• Extremely detailed Reference Sheets completely covering the technologies – these Reference
Sheets include the necessary syntax and examples to have you mastering configurations on Cisco
devices quickly and easily.
• Video-On-Demand recordings by NetMasterClass instructors – these videos provide walkthrough
configurations for the most important features on actual Cisco equipment.
• Practice Exam Database – prepare thoroughly and completely for the Cisco Certified Voice
Professional (CCVP) exam number 642-432 (CVOICE) using our online testing simulation tool.

This course is recommended for the following IT Professionals:

• Those that want to master Cisco Voice over IP for network implementations
• Those that want to obtain the CCVP® Certification
• Those that want to begin CCIE® Voice track preparation

Disclaimer
NetMasterClass, LLC is an independent training and consulting company based in Herndon, Virginia. The
terms “Cisco”, “Cisco Systems” and “CCIE” are the trademarks of Cisco Systems, Inc. NetMasterClass,
LLC is Cisco Learning Partner.

You agree that the information you purchase is protected by copyright and is for YOUR USE ONLY. The
Video-on-Demand files, quiz questions and this PDF file CAN NOT be resold, reproduced, transmitted,
transcribed in any form or by any means electronically, mechanically or otherwise.

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Cisco Voice over IP

I. Telephony Fundamentals
a. Traditional Telephony Overview
i. Telephone Network Components
1. Telephone – analog or digital
2. Private Branch Exchange (PBX) – privately owned telephone switch
3. Key Telephone System – privately owned telephone switch used in
smaller environments (i.e. less than 50 users)
4. CO Switch – telephone switched owned and operated by the telephone
company
5. Local Loop – a pair of wires (tip and ring) that connects a telephone in a
residence with a CO switch
6. Interoffice Trunk – interconnects two CO switches located in different
offices
7. Tie Trunk – interconnects two PBXs
8. CO Trunk – interconnects a PBX and a CO switch
b. Analog Voice Concepts
i. Analog Signaling Categories
1. Supervisory Signaling
a. On-Hook
b. Off-Hook
c. Ringing
2. Address Signaling
a. Pulse - rapidly opens and closes the local loop
b. Dual-Tone Multifrequency (DTMF) – generates two simultaneous
frequencies
3. Information Signaling
a. Dial Tone
b. Ringback
c. Reorder
d. No Such Number
e. Busy
f. Congestion
g. Phone Off-Hook
ii. Analog Trunk Signaling Types
1. Loop-Start
a. Seizes a line when loop current is detected
b. Can suffer from glare (i.e. picking up a handset to place a call
before the phone rings and being connected to an incoming
caller)
2. Ground-Start
a. Seizes a line when the ring lead has a ground potential
b. Does not suffer from glare
3. E&M Wink Start Signaling – the most common E&M signaling type
4. E&M Immediate-Start Signaling
5. E&M Delay Start
iii. Echo on Analog Circuits
1. Characteristics of Echo

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a. Typically due to an impedance mismatch in a two-wire to four-

wire hybrid circuit
b. Becomes more noticeable when the echo waveform is louder and
the delay between the original waveform and the echo waveform
is longer
2. Ways to Combat Echo
a. Echo Suppression – creates a unidirectional speech path, where
the waveform with the greatest amplitude (i.e. volume) is
permitted
b. Echo Cancellation – recognizes an echo waveform and cancels
the waveform by superimposing a copy of the echo waveform
which has been phase shifted 180 degrees
c. Digital Signaling
i. Digital Circuits
1. T1 – contains 24 channels which can each be used as a voice path
2. E1 – contains 32channels, 30 of which can be used as a voice path, with
one channel being used for framing and synchronization and another
channel being used for signaling
ii. Signaling Strategies
1. Channel Associated Signaling (CAS) – sends signaling as part of the
voice path
a. Uses the framing bit for every sixth frame in a T1 superframe (SF)
or extended superframe (ESF) to carry signaling (i.e. “robbed-bit
signaling”)
b. Uses bits in the seventeenth time slot inside an E1 multiframe
(i.e. “timeslot 16” or “TS16” since the timeslot numbering begins
at 0) to carry signaling information for the 30 channels than can
carry voice
2. Common Channel Signaling (CCS) – uses a signaling protocol sent in a
channel separate from voice
a. Typically uses the last channel in a T1 circuit to carry the
signaling protocol, such as Q.931
b. Typically uses TS16 to carry a signaling protocol, such as Q.931
iii. Integrated Services Digital Network (ISDN) Circuits
1. Basic Rate Interface (BRI) – contains two bearer channels, which can
carry voice, data, and/or video and one delta channel, which carries the
Q.931 protocol, commonly written as “2B-1D”
2. Primary Rate Interface (PRI) – based on a T1 (23B-1D) or an E1 (30B-
1D) circuit
iv. Q Signaling (QSIG) – a signaling protocol commonly used to communicate
between different vendors’ PBXs
v. Signaling System 7 (SS7) – a signaling protocol commonly used to communicate
between CO switches
II. Voice over IP (VoIP) Fundamentals
a. Converting Analog Waveforms to VoIP Packets
i. Sampling
1. Nyquist Theorem – sample at a rate that is at least twice the highest
frequency being sampled
2. For voice, 4,000 Hz X 2 = 8,000 samples per second
ii. Quantization

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1. Assigning a numeric value to the sample, based on the amplitude and

polarity of the sample
2. Quantization Error – results in audible noise and hiss due to the rounding
off a sample’s actual amplitude to the nearest discrete value on the
quantization scale
3. Logarithmic Quantization
a. Uses a logarithmic scale, as opposed to a linear scale
b. Logarithmic scale divided into segments
c. Segments divided into steps
d. Results in more accurate measurements (i.e. step values) being
taken at lower amplitudes (i.e. volumes), thus reducing the noise
and hiss resulting from quantization error
e. Types of Logarithmic Quantization
i. a-law – used in most countries
ii. mu-law – used in the U.S., Canada, and Japan
iii. Encoding
1. Representing the sample value in binary
2. 8 bit sample size – 1 polarity bit, 3 segment bits, and 4 step bits
iv. Compression (optional) – taking the binary values representing sampled voice
and compressing those values in order to reduce the WAN bandwidth demand
b. VoIP Introduction
i. Roles of VoIP
1. Signaling – protocols reside at Layer 5 (i.e. the Session Layer) of the OSI
Model
a. H.323 - a mature peer-to-peer gateway control protocol
b. SIP (Session Initiation Protocol) – a newer peer-to-peer gateway
control protocol which is based on open standards
c. MGCP (Media Gateway Control Protocol) - a client-server
gateway control protocol originally developed by Cisco
d. Megaco/H.248 – an enhanced version of MGCP
e. SCCP (Skinny Client Control Protocol) – typically used for
signaling between a Cisco IP Phone and Unified CallManager
2. Database
a. Billing
b. Caller ID
3. Call control
a. Call connect
b. Call disconnect
4. Codecs
a. G.711 – requires 64 kbps of bandwidth (payload only)
b. G.729 – requires 8 kbps of bandwidth (payload only)
c. G.729a – requires 8 kbps of bandwidth (payload only), but is less
processor intensive (i.e. less “complex”) than G.729
d. G.729b – requires 8 kbps of bandwidth (payload only), but adds
comfort noise generation (CNG) and voice activity detection
(VAD) to G.729
e. G.729ab – requires 8 kbps of bandwidth (payload only), but is
less processor intensive than G.729 and adds CNG and VAD
features to G.729
c. VoIP Network Elements

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i. IP Phone – in some cases, powered via in-line power from Cisco Catalyst
switches, and typically obtains an IP address from a DHCP server
ii. Gatekeeper – keeps track of bandwidth available on the IP WAN and maintains a
database of gateway IP addresses that can be used to reach certain phone
numbers
iii. Gateway – a router or a switch that takes VoIP packets and sends them into a
traditional analog or digital environment
iv. Multipoint Control Unit (MCU) – contains DSP resources supporting conference
calls
v. Call Agent – the call routing component of a VoIP network (e.g. Cisco Unified
CallManager)
vi. Application Server – runs an application (e.g. Cisco Unity) to add features to the
network
vii. Video Conferencing Stations – for example, PCs running the Microsoft
NetMeeting or Cisco Unified Video Advantage applications
d. VoIP Prerequisites
i. Characteristics of a VoIP Network
1. Connectionless – resulting in dropped voice packets not being
retransmitted
2. Can transmit traffic across multiple paths – can cause voice packets to
arrive out of sequence at the destination router
3. Not as government regulated, as compared to traditional telephony
service
ii. Quality of Service (QoS) – to ensure real-time delivery of voice packets
1. QoS Issues
a. Jitter – variation in packet arrival times
b. Delay – the overall delay between the time the speaker speaks
and the listener hears
c. Packet Loss – the dropping of packets (e.g. discarding packets
due to congestion)
2. QoS Mechanisms
a. Classification and Marking
i. Recognizes a packet’s traffic type, categorizes the packet
into a predefined class, and marks the packet (by altering
bits in the packet) so that it can be easily recognized by
the next-hop device
ii. Recommended markings
1. Voice – DSCP value of EF (Expedited
Forwarding)
2. Call control – DSCP value of AF31 (Assured
Forwarding 31) NOTE: Cisco’s new
recommendation is for call control traffic to be
marked with a DSCP value of CS3 (Class
Selector 3); however, for purposes of the
CVOICE exam, use the previous
recommendation of AF31
b. Congestion Management
i. Storing packets in an interface’s output queue and
forwarding those packets out of the queue as bandwidth
becomes available, based on the priority of the packet

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ii. Modern queuing mechanisms

1. Class-Based Weighted Fair Queuing (CB-WFQ)
a. Places traffic into different classes and
assures that each class of traffic can
have at least a specified amount of
bandwidth during times of congestion
b. Does not provide a priority queue for
voice
2. Low Latency Queuing (LLQ)
a. Similar to CB-WFQ, with the addition of a
priority queue
b. Voice is typically placed in the priority
queue, which is emptied ahead of other
queues, up to a certain bandwidth limit
(to prevent the starvation of other traffic
types)
c. Congestion Avoidance – prevents an interface’s output queue
from ever filling to capacity by discarding packets more and more
aggressively as the queue begins to fill
d. Traffic Conditioning – uses policing and/or shaping tools to limit
the bandwidth consumed by a particular traffic type
e. Link Efficiency – attempts to make a more efficient use of limited
WAN bandwidth through compression and link fragmentation and
interleaving (LFI)
e. Gateway Overview
i. Gateway – a router or switch that interconnects two types of environments (e.g. a
PBX using E&M Wink Start signaling and an IP telephony network using H.323
signaling)
ii. Gateway Selection Criteria
1. Analog or digital
2. Gateway capacity
3. Interface types supported
4. Call control protocol supported
5. Voice compression type (if any)
6. DID (Direct Inward Dial), CLID (Calling Line ID), and modem/fax relay
requirements
7. Inline power support (if needed)
8. Survivable Remote Site Telephony (SRST) support (if needed)
9. Country in which the gateway will be located
iii. Gateway Connections – typically connects to the PSTN for off-net calls and to a
Unified CallManager (UCM) server for on-net calls
iv. Service Provider Gateway Requirements
1. SS7 compatibility
2. High-availability and QoS
3. Scalability
f. Voice Packets
i. Real-time Transport Protocol (RTP)
1. Carries voice traffic using, by default on Cisco equipment, UDP port
numbers 16,384 – 32, 767
2. Resides at Layer 4 of the OSI Model

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ii. RTP Header Compression (cRTP)

1. Uncompressed combined IP/UDP/RTP header – 40 bytes
2. Compressed header – 2 bytes without UDP checksums and 4 bytes with
UDP checksums
iii. RTP Control Protocol (RTCP) – monitors the quality of an RTP stream
g. VoIP Design Considerations
i. Deployment Models
1. Centralized Deployment Model
a. Centrally located call routing intelligence (e.g. a cluster of Unified
CallManager clusters located at a company’s headquarters)
b. MGCP or Megaco/H.248 appropriate for call control
c. Scales to 10,000 IP Phones (with an MCS 7835 Unified
CallManager platform) or 30,000 IP Phones (with an MCS 7845
Unified CallManager platform)
d. Uses Survivable Remote Site Telephony (SRST) to maintain
service for remote sites during a WAN outage
2. Distributed Deployment Model
a. Call routing intelligence located at all locations (e.g. each site
having its own cluster of Unified CallManager servers)
b. H.323 or SIP appropriate for call control
c. Can use a gatekeeper for scalability
ii. Factors Impacting Voice Quality
1. Bandwidth availability
2. Echo
3. Delay and delay variation (i.e. jitter)
4. Playout delay buffer
iii. Ways to Mitigate Quality Issues Using QoS Mechanisms
1. Allocating specific bandwidth amounts for defined traffic classes
2. Dropping data packets, as opposed to voice packets, during times of
congestion
3. Treating voice packets with higher priority (e.g. transmitting voice packets
before transmitting data packets)
4. Shaping traffic to prevent oversubscription of a WAN link
5. Using priority markings to identify voice traffic as high priority traffic
iv. Traffic Engineering
1. Goal – provision sufficient bandwidth to accommodate a certain
percentage of calls during the busiest hour of the day (i.e. “Busy Hour
Traffic”)
2. Grade of Service (GoS) – an acceptable percentage of voice calls to be
blocked during the busiest hour of the day, typically 1 percent, written as
“P(.01)”
3. Offered Traffic – the amount of total phone use during the busiest our of
the day, measured in Erlangs, where one Erlang equals:
a. 60 call minutes
b. 3600 call seconds
c. 36 centum call seconds
4. Number of Trunks Required – calculated from a table (e.g. an Erlang B
table) or calculator, when the GoS and offered traffic is known

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5. Bandwidth Required – calculated based on the number of trunks required,

and based on the VoIP codec, payload size, Layer 2 transport, whether or
not compression is being used, and any additional overhead
a. Per Call Required Bandwidth = (Total Packet Size x Payload-only
Bandwidth Requirement) / Payload Size
i. Total Packet Size = Layer 2 Overhead + IP/UDP/RTP
Headers + Payload
1. Layer 2 Overhead
a. PPP – 6 bytes
b. Frame Relay – 6 bytes
c. Ethernet - 18 bytes
2. IP/UDP/RTP Headers - 40 bytes
3. Payload
a. G.729 – 20 bytes
b. G.711 – 160 bytes
ii. Payload-only Bandwidth Requirement
1. G.729 – 8 kbps
2. G.711 – 64 kbps
b. Example
i. Codec = G.729
Layer 2 Transport = Frame Relay
Per Call Required Bandwidth = ((6 + 40 + 20) bytes x 8
kbps) / 20 bytes = 26.4 kbps
v. Security Considerations
1. Must support required VoIP communications protocols (e.g. using a
stateful firewall to dynamically determine the UDP port numbers selected
for a VoIP stream)
2. Must secure the transport of VoIP traffic (e.g. sending voice packets over
an IPsec-protected virtual private network (VPN))
3. Must not interfere with the existing data network’s security (e.g. not open
up any ports on a firewall which could be used to compromise the
underlying data network)
h. Numbering Plan
i. A component of a dial plan, which specifies the format of numbers to be dialed
ii. Other components of a dial plan:
1. Rules dictating which paths should be used to place specific calls (e.g.
least cost routing)
2. Rules specifying which phones are allowed to call specific destinations
(e.g. disallowing a lobby phone from calling an international number)
iii. Scalable Numbering Plans
1. Example – North American Numbering Plan (NANP), which uses the
NXX-NXX-XXXX (where N is a number in the range 2 - 9) pattern for long
distance calls
2. Characteristics of a Scalable Numbering Plan:
a. Logic distribution
b. Hierarchical design
c. Simplified provisioning
d. Reduced post-dial delay
e. Redundancy
3. Potential Numbering Plan Integration Issues

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a. Phone numbers of varying lengths

b. International dialing
c. Overlapping numbering plans (e.g. when merging two companies
together)
4. 911 Terminology
a. ANI (Automatic Number Configuration) – the phone number of the
person placing the call (i.e. the “calling party”)
b. Automatic Location Identification (ALI) – a database that
associates a phone number with a physical location
c. PSAP (Public Safety Answering Point) – where a 911 call is
terminated
d. ERL (Emergency Response Location) – used in mobile
environments to identify the appropriate location from which an
emergency call was placed (e.g. a specific floor of a building)
e. ELIN (Emergency Location Identification Number) – used in
mobile environments, where the ELIN phone number replaces the
ANI phone number when an emergency call is sent to the PSAP
f. MSAG (Master Street Address Guide) – a government-
maintained database that maps geographic regions to the PSAPs
that are responsible for handing calls coming from those regions
g. Selective Router – A phone switch that routes 911 calls to
appropriate PSAPs, based on a call’s ANI information
h. CAMA (Centralized Automated Message Accounting) – an analog
trunk that connects a customer’s phone switch directly to a
selective router
i. CER (Cisco Emergency Responder) – a Cisco application that
works with Unified CallManager to support location updates,
including support for mobile environments
III. VoIP Applications
a. VoIP Optional Services
i. Hospitality
1. Cisco Building Broadband Service Manager (BBSM)
2. Cisco IP Phone
3. Cisco Content Transformation Engine (CTE)
ii. IP Centrex
iii. Multitenant
1. Multiple Dwelling Units (MDUs)
2. Multiple Tennant Units (MTUs)
iv. Prepaid Calling Card
v. Collaborative Computing
1. Reduces travel expenses
2. Transparent to the end user
3. Real-time interaction
4. Document and calendar sharing
vi. Voice Enabled Web Applications (VXML)
vii. Unified Messaging
viii. Hoot and Holler – an always-on multiuser conference, typically used by brokerage
firms
ix. Toll Bypass
b. Fax and Modem Considerations

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i. VoIP Fax Options

1. Fax Pass-Through – uses G.711 as the codec, which doesn’t compress
the fax tones
2. Cisco Fax Relay – uses a Cisco proprietary approach to send fax tones
across a link in specially marked RTP segments
3. T.38 – an industry-standard approach similar to Cisco Fax Relay
4. T.37 – a store-and-forward approach that allows a router acting as an “on-
ramp” gateway to receive a fax, convert the received fax into an e-mail
attachment, and e-mail the attachment to an SMTP server, which can
then e-mail the attachment to a router acting as an “off-ramp” gateway,
which can take the e-mail attachment, convert it back into a fax format,
and send the fax to the destination fax machine
ii. VoIP Modem Options
1. Modem Pass-Through – uses G.711 to send modem tones
uncompressed across a VoIP network
2. Modem Relay – uses the Simple Packet Relay Transport (SPRT) protocol
to transmit modem tones
c. Cisco VoIP Applications
i. Cisco Unified CallManager – the call processing component of Cisco’s IP
telephony solution; multiple Unified CallManager servers can be grouped together
in a Unified CallManager redundancy group
ii. Cisco Unified CallManager Express – an IOS feature that allows Cisco IP Phones
to register with a router and provides call processing functions for smaller
environments (i.e. up to 250 users)
iii. Cisco Unity – a unified messaging application that allows fax messages, e-mail
messages, and voice mail messages to be stored in a single repository
iv. Cisco IP Contact Center (IPCC) Express – a call center application that can run
on a single server
v. Cisco Unified Video Advantage – a hardware (i.e. a USB camera) and software
product that supports video (displayed on a PC) to accompany an audio call
(heard over a Cisco IP Phone)
d. Virtual Meeting Solutions
i. Cisco MeetingPlace – Cisco’s conferencing application that integrates voice and
web conferencing
ii. Cisco IP Videoconferencing – uses dedicated video-conference appliances to mix
audio and video streams together in a conference
IV. Voice Port Configuration
a. Call Categories
i. Local calls – calling and called parties both attached to the same router
ii. On-net call – calling and called parties attached to different routers which reside
on the same network
iii. Off-net call – a call where the calling party is on the network, while the called party
is not on the network (e.g. on the Public Switched Telephone Network (PSTN))
iv. Private Line Automatic Ringdown (PLAR) – occurs when a phone’s handset is
picked up, and the router automatically connects the phone to a preconfigured
number, without the calling party dialing any digits
v. PBX-to-PBX call – occurs when both the calling and called parties are behind their
own local PBX, and the PBXs are connected to different routers, which are
interconnected over an IP network

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vi. Cisco Unified CallManager to Cisco Unified CallManager call – calling and called
parties are both IP Phones and connected to different Unified CallManager
clusters, and the clusters communicate over a trunk (e.g. gatekeeper-controlled
intercluster trunk)
vii. On-net to off-net call – a call that was intended to be on-net, but was diverted off-
net (e.g. over the PSTN) due to the IP WAN being unavailable
b. Foreign Exchange Station (FXS) Ports
i. Tunable FXS parameters:
1. signal - loopstart or groundstart
2. cptone – determines how the call progress tones sound
3. description – allows an administrator to document what the port connects
to
4. ring frequency – frequency of the ringing voltage, used for compatibility
with older mechanical ringers
5. ring cadence – the ringing pattern, for example two seconds on and four
seconds off
6. disconnect-ack – removes line power if the phone goes on-hook
7. busyout – takes the line out of service
8. station id name/number – specifies caller ID information associated with a
port
ii. Example:
1. voice-port 1/0/1
signal loopstart
cptone US
ring cadence pattern01
c. Foreign Exchange Office (FXO) Ports
i. Tunable FXO Parameters:
1. signal – loopstart or groundstart
2. ring number – the number of incoming rings before the FXO port answers
3. dial-type – pulse or dtmf
4. description – allows an administrator to document what the port connects
to
5. supervisory disconnect – allows a phone switch (e.g. a PBX) to signal the
port to disconnect
ii. Example:
1. voice-port 1/1/1
signal groundstart
ring number 2
dial-type dtmf
d. Ear and Mouth (E&M) Ports
i. Tunable E&M Parameters:
1. signal – wink-start, immediate, or delay-dial
2. operation – the number of tip and ring wires used to carry voice (i.e. 2-
wire or 4-wire)
3. type – 1, 2, 3, or 5 (NOTE: Cisco does not support E&M Type 4)
4. auto-cut-through – used when connecting to a PBX that doesn’t provide
an M-lead response (i.e. applying a -48 V potential to the M-lead to
indicate a line seizure)
5. description – allows an administrator to document what the port connects
to

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ii. Example:
1. voice-port 2/1/1
signal wink-start
operation 4-wire
type 2
e. Timers
i. Timer Parameters
1. timeouts initial – the amount of time dial-tone is present after you first pick
up the handset
2. timeouts interdigit – the maximum amount of time allowed between dialed
digits
3. timeouts ringing – the maximum amount of time that ringing voltage will
be sent to a phone
4. timing digit – the duration of a dialed DTMF digit
5. timing interdigit – the duration of the silence between dialed DTMF digits
6. timing hookflash-in/hookflash-out – used to adjust the “flash” behavior
(e.g. used in a call waiting environment to put one call on hold while
picking up another incoming call)
ii. Example:
1. voice-port 2/0/0
timeouts initial 20
timeouts interdigit 20
timing hookflash-in 500
timeouts ringing 120
f. Digital Controllers
i. T1 Controller Parameters
1. Framing
a. SF (Super Frame) – a group of twelve standard 193-bit T1 frames
b. ESF (Extended Superframe) – a group of 24 standard 193-bit T1
frames
2. Linecoding
a. AMI (Alternate Mark Inversion)
b. B8ZS (Bipolar Eight Zero Substitution)
3. Clocking
a. Line – from the network
b. Internal – from the router
ii. E1 Controller Parameters
1. Framing
a. CRC4
b. no-CRC4
c. Australia
2. Linecoding
a. AMI
b. HDB3
3. Clocking
a. Line
b. Internal
iii. DS0 Groups
1. Grouping of channels on a digital circuit configured for a common
signaling type

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2. ds-group group_number timeslots range_of_timeslots type

signaling_type
iv. Example:
1. controller t1 1/0
framing esf
linecode b8zs
clock source line
ds0-group 1 timeslots 1-8 type fxs-ground-start
ds0-group 2 timeslots 9-16 type fxo-loop-start
ds0-group 3 timeslots 17-24 type e&m-wink-start
g. ISDN Interfaces
i. Configuration Steps
1. Specify ISDN switch-type (or optionally Q Signaling (QSIG))
2. Create a PRI-group
3. Configure the D channel (i.e. the signaling channel) to direct calls to
DSPs (as opposed to, for example, modems)
ii. Example:
1. isdn switch-type primary-5ess
controller t1 1/0
pri-group timeslots 1-23
interface serial 1/0:23
isdn incoming-voice voice
h. Transparent Common Channel Signaling (T-CCS)
i. Allows proprietary signaling information to transparently pass through a channel
on a digital circuit
ii. The ds0-group 1 timeslots 24 type ext-sig command configures channel 24 for
external signaling
iii. The codec clear-channel command allows the signaling to pass through the
DSP without compression or processing
i. Verification and Troubleshooting Commands
i. show voice port
ii. show voice port port_identifier
iii. show voice port summary
iv. show voice busyout
v. show voice dsp
vi. show controller {T1 | E1} controller_identifier
vii. show isdn status
viii. debug isdn q921
ix. debug isdn q931
j. Voice Quality Tuning Commands
i. input gain gain_in_db (where the gain_in_db ranges from -6 through 14)
ii. output attenuation attenuation_in_db (where the attenuation_in_db ranges
from -6 through 14)
iii. impedance {600c | 600r | 900c | 900r | complex1 | complex2}
iv. no echo-cancel enable (to disable echo cancellation, which is enabled by
default)
v. echo-cancel coverage time (where time is the time in milliseconds that the voice
port retains (i.e. “remembers”) waveforms used for echo detection)
vi. non-linear (suppresses all transmission from a phone from which speech is not
currently being detected, which can lead to clipping as speaking begins)

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vii. Example:
1. voice-port 1/1/1
input gain 2
output attenuation 1
echo-cancel coverage 24
impedance 600r
V. Dial Peer Configuration
a. Dial Peers
i. Addressable endpoints
ii. VoIP
1. Points to a remote IP address
2. Example:
a. dial-peer voice 1 voip
destination-pattern 1500
session target ipv4:10.1.2.3
iii. POTS
1. Points to a local port
2. Example:
a. dial-peer voice 2 pots
destination-pattern 2000
port 1/1/1
iv. Destination-Pattern Wildcards
1. Plus sign “+” - indicates a standard E.164 (where E.164 represents the
ITU-T International Public Telecommunications Numbering Plan) phone
number
2. Period “.” – matches a single dialed digit
3. Brackets “[ ]” – matches a range of numbers in the brackets (e.g. 311[0 –
5] matches numbers in the range 3110 through 3115
4. “T” – represents a variable-length dial string, which a caller can terminate
with the pound (i.e. “#”) key
5. Example 1 – Create a VoIP dial peer to match four-digit phone numbers
at a remote site, that begin with a 4:
dial-peer voice 4 voip
destination-pattern 4…
session target ipv4:10.2.2.2
6. Example 2 – Create a POTS dial peer to point out to the PSTN (i.e. with a
variable-length dial string), where the PSTN access code is 9:
dial-peer voice 4 pots
destination-pattern 9T
port 1/1/0
b. Call Legs
i. Incoming
1. The stage of a call as it comes into a router
2. Matches an inbound dial peer
a. incoming called-number – matches a dial peer based on the
DNIS (Dialed Number Information Service) information
b. answer-address – matches a dial peer based on the ANI
(Automatic Number Identification) information
c. destination-pattern – uses the caller ID information (i.e. the ANI)
to match the incoming call leg

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d. port – matches the port associated with the incoming call leg
3. Uses Dial Peer 0 (i.e. the “Default Dial Peer”), when no inbound dial peer
is matched
ii. Outgoing
1. The stage of a call as it leaves a router
2. Matches an outbound dial peer
a. Outbound dial peer matched based on the destination-pattern
command
b. Most specific match used if multiple dial peers have a matching
destination
c. Uses the preference value command (where value is a number
in the range of 0 – 10, with lower numbers being more preferable)
to select a dial peer if multiple dial peers have equally specific
destination patterns (NOTE: The default preference value for a
dial peer is 0.)
c. Digit Processing
i. Digit Forwarding on a POTS Dial Peer
1. Only digits matched by a wildcard are forwarded, by default
2. Example
a. dial-peer voice 1000 pots
destination-pattern 555….
port 1/1/1
b. Dialed Digits = 5551234
c. Forwarded Digits = 1234
ii. Digit Analysis
1. Digits collected one at a time until a dial peer is matched, after which the
call is setup
2. Example
a. dial-peer voice 100 voip
destination-pattern 555
session target ipv4:10.1.1.1
dial-peer voice 200 voip
destination-pattern 5551234
session target ipv4:10.2.2.2
b. Caller attempts to dial 5551234, in the above example, but the
initial “555” in the dial string causes dial peer 100 to be matched
instead of dial peer 200
d. Digit Manipulation Commands
i. prefix – adds specified digits to the dial string before the dial string is forwarded
out the telephony interface
ii. forward-digits – specifies how many digits are forwarded out the telephony
interface, regardless of how many numbers were explicitly matched and how
many numbers were matched by a wildcard
iii. num-exp – replaces one number with another number (e.g. a telecommuter could
be assigned a four-digit directory number of 2020, which mapped, using the
command num-exp 2020 5552020, to a fully-qualified PSTN number of 555-
2020)
iv. translation-rule – performs digit translation to a specific dial peer, rather than
globally as the num-exp command does
1. Example:

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a. A translation rule that takes a number beginning with a 2 and

prepends 859555 to the dialed number
b. translation-rule 2
rule 1 ^2. 8595552
c. Syntax Explanation
i. translation-rule 2 – enters translation rule configuration
mode
ii. rule 1 ^2. 8595552
1. The caret symbol and the period cause the rule
to match a dial string beginning with a 2 followed
by one or more digits
2. The replacement string will replace the explicitly
matched digit, which is a 2 in this case
3. The result is that any dial string of at least two
digits that begins with a 2, will have the leading 2
replaced with 8595552, and the resulting dial
string will be 8595552 followed by the remaining
digits in the original dial string that were not
explicitly matched
iii. Verification
1. Test command - test translation-rule 2 2020
2. Result – The replaced number is 8595552020
e. Connection Types
i. PLAR (Private Line Automatic Ringdown)
1. Associates a voice port with a dial peer, such that a preconfigured
number is dialed when the voice port goes off-hook
2. connection plar dial_string
ii. PLAR-OPX (Private Line Automatic Ringdown – Off-Premise Extension)
1. Used to make off-site extensions appear as if they were connected to a
local PBX
2. connection plar-opx dial_string
iii. Trunk
1. Creates a permanent connection between two PBXs, over an IP WAN
2. connection trunk dial_string [answer-mode], where answer-mode
tells the router to wait for an incoming call before establishing a trunk
iv. Tie-Line
1. Creates an on-demand connection between two PBXs, over an IP WAN
2. connection tie-line dial_string
VI. Call Control Protocols
a. H.323
i. H.323 Characteristics
1. Peer-to-peer protocol
2. Mature protocol
3. Sends messages using Abstract Syntax Notation-1 (ASN.1), as opposed
to plain text
4. An ITU-T standard
ii. H.323 Protocols
1. H.225 – performs call setup between two H.323 endpoints, using TCP
2. H.245 – performs a capabilities exchange and opens up a logical channel
between two H.323 endpoints using TCP

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3. H.225 RAS (Registration Admission and Status) – communicates with an

H.323 gatekeeper using UDP
iii. H.323 Components
1. Terminal
a. The H.323 device that interfaces with a user (e.g. an IP phone
which supports H.323 or a PC running an H.323 application)
b. Must support the G.711 codec
2. Gateway
a. Converts between two audio formats and signaling types (e.g.
between the PSTN and a VoIP network)
b. IP-to-IP Gateway – supports the interconnection of two VoIP
networks
3. Gatekeeper
a. Required Features
i. Address translation – coverts between a phone number
and an IP address which can be used to reach the phone
number
ii. Admission control – granting or denying admission to the
network, based on available bandwidth
iii. Bandwidth control – manages endpoint bandwidth
requirements (e.g. allows mid-call changes in bandwidth)
iv. Zone management – allows H.332 endpoints and
gateways to register with a specific zone association
b. Optional Features
i. Call control signaling – the ability to forward the signaling
protocols (i.e. H.225 and H.245) between the H.323
endpoints or gateways setting up the call, as opposed to
having the signaling protocols to travel directly between
the H.323 endpoints or gateways
ii. Call authorization – can reject a call due to an
authorization failure
iii. Bandwidth management – can limit the number of
simultaneous connections to an IP network, based on
network resources (i.e. provides Call Admission Control
(CAC))
iv. Call management – maintains call records
4. Multipoint Control Unit (MCU)
a. Supports conference calls
b. Incorporates a multipoint controller and one or more multipoint
processors
iv. H.323 Call Establishment
1. Combinations of endpoint/gateway and gatekeeper call establishment
a. Endpoint/gateway to endpoint/gateway
b. Endpoint/gateway to gatekeeper
c. Gatekeeper to gatekeeper
v. H.323 RAS Request Messages
1. Gatekeeper discovery (GRQ) – locates a gatekeeper, via either a
preconfigured unicast address or multicast discovery
2. Terminal and gateway registration (RRQ) – an endpoint or gateway
registers itself with the discovered gatekeeper

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3. Terminal and gateway unregistration (URQ) – an endpoint or gateway

unregisters itself with a gatekeeper
4. Location request (LRQ) – a message sent from one gatekeeper to
another gatekeeper to resolve the IP address of an endpoint or gateway
located in a remote zone
5. Call admission (ARQ) – a message sent by an endpoint or gateway to
request permission from the gatekeeper to place a call across the network
6. Bandwidth change (BRQ) – requests a mid-call adjustment in the
bandwidth allocated for a call
7. Disengage (DRQ) – a message sent when a call disconnects
8. Status queries (IRQ) – a query sent from the gatekeeper to determine the
current status of an endpoint or gateway
vi. H.323 Call Setup
1. Basic Call Setup
a. H.225 performs the initial call setup request
b. H.245 performs a capabilities exchange and opens a logical
channel
c. bidirectional RTP flows between the H.323 endpoints/gateways
2. H.323 Fast Connect – communicates all of the information necessary to
setup a call within a single exchange between the H.323
endpoints/gateways
3. Single Gatekeeper
a. All H.323 endpoints/gateways register with a single gatekeeper
b. The H.323 endpoints/gateways register as members of specifics
zones
c. Since all H.323 endpoints/gateways register with the same
gatekeeper, from the perspective of that gatekeeper, all zones
are considered to be local zones
d. The gatekeeper permits or denies call attempts between
endpoints/gateways in different zones
4. Multiple Gatekeepers
a. Each H.323 endpoint/gateway registers with one of multiple
gatekeepers
b. The H.323 endpoints/gateways register as members of specifics
zones
c. From the perspective of a gatekeeper, H.323 endpoints/gateways
that register with a different gatekeeper are considered to be in
remote zones
d. When an admission request (ARQ) is received by a gatekeeper
for a remote zone, that gatekeeper queries a different gatekeeper
via an LRQ to determine the IP address of the destination H.323
endpoint/gateway with which the call should be setup
5. Proxy Server
a. Diverts call setup through one or more proxy servers, as opposed
to through one or more gatekeepers
b. Can be used to establish an optimal path for signaling traffic
through the network
c. Can be used to allow an H.323 flow to pass through a firewall
d. Can help secure a network by concealing the IP addresses of the
H.323 endpoints/gateways involved in the call setup

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vii. H.323 Redundancy

1. HSRP
2. VRRP
3. Configuring a gateway with two gatekeepers or for multicast discovery of
a gatekeeper
4. Configuring multiple gatekeepers with the same prefix
5. Configuring multiple gateways with the same prefix
viii. H.323 Configuration
1. Gateway Syntax
a. gateway – enables the gateway process on a router
b. h323-gateway voip interface – declares that an interface is an
H.323 interface
c. h323-gateway voip id Zone_Name IP_Address_of
_Gatekeeper – points the gateway to the IP address of a
gatekeeper and specifies the zone with which the gateway wishes
to register (NOTE: the default port number of 1719 will be entered
automatically)
d. h323-gateway voip h323-id – identifies the name by which the
gateway will be identified when viewing the output of show
commands
e. h323-gateway voip tech-prefix prefix – identifies the type of
calls this gateway accepts (e.g. voice calls, video calls, modem
calls, fax calls, or default technology calls where no technology
prefix was specified in the dial string)
f. session target ras – issued in dial peer configuration mode, this
command specifies that a dial peer should consult the
gatekeeper, via the RAS channel, to determine the destination IP
address to which a call setup message should be sent
2. Gatekeeper Syntax
a. gatekeeper – enables the gatekeeper process on a router
b. zone local Zone_Name Domain_Name RAS_IP_Address –
H.323 endpoints/gateways can register with the gatekeeper as
members of the specified zone by sending an RRQ RAS
message to the specified IP address (NOTE: Only one zone
local command can contain an IP address)
c. zone prefix Zone_Name Phone_Number – specifies one or
more (through the use of wildcards) phone numbers that are
reachable in a specific zone
d. gw-type-prefix Technology_Prefix default-technology –
specifies the technology prefix which is considered to be the
“default technology prefix,” and calls that do not specify a
technology prefix in their dial string can be routed to H.323
endpoints/gateways that have registered with the default
technology prefix
e. no shutdown – brings the gatekeeper online
3. Example – H.323 Gateway
a. gateway
!
interface serial 0/0
ip address 10.1.1.1 255.255.255.0

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h323-gateway voip interface

h323-gateway voip id ZoneA ipaddr 10.1.1.2 1719
h323-gateway voip h323-id R1
h323-gateway voip tech-prefix 1#
!
dial-peer voice 100 voip
destination-pattern 5…
session target ras
4. Example – H.323 Gatekeeper
a. gatekeeper
zone local ZoneA domain.com 10.1.1.2
zone local ZoneB domain.com
zone prefix ZoneA 4…
zone prefix ZoneB 5…
gw-type-prefix 1#* default-technology
no shutdown
ix. H.323 Verification and Troubleshooting Commands
1. show call active voice [brief]
2. show call history voice
3. show gateway
4. show gatekeeper calls
5. show gatekeeper endpoints
6. show gatekeeper gw-type-prefix
7. show gatekeeper zone status
8. show gatekeeper zone prefix
9. show gatekeeper zone status
10. debug voip ccapi inout (NOTE: debug commands provide real-time
information, as opposed to show commands)
b. SIP
i. SIP Characteristics
1. Peer-to-peer protocol
2. An IETF standard
3. Uses the concept of inviting a participant into a session (using an INVITE
message)
4. Based on open standards
5. Sends messages in plain text
6. Addresses resemble an e-mail address (e.g.
sip:[email protected]; user=phone)
ii. SIP Components
1. User Agent Client (UAC) – initiates a SIP call
2. User Agent Server (UAS) – the recipient of a SIP call
3. Proxy Server – performs the call setup on behalf of the UAC
4. Redirect Server – informs the UAC the next server to contact
5. Registrar Server – registers the location of user agents and keeps track of
what phone numbers are available via those user agents
6. Location Server – performs address resolution for SIP proxy and redirect
servers
iii. SIP Messages
1. Requests – a SIP messages sent from a UAC to a UAS
a. INVITE – invites a participant into a session

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b. ACK – acknowledges the receipt of an INVITE message

c. BYE – terminates a SIP call
d. CANCEL – does not end current sessions, but rather cancels
pending searches
e. OPTIONS – queries a SIP server for its capabilities
f. REGISTER – registers a SIP user agent with a registrar server
2. Responses – a SIP messages sent from a UAS to a UAC
a. 1XX – informational response
b. 2XX – successful response
c. 3XX – redirection response
d. 4XX – client error response
e. 5XX – server error response
f. 6XX – global failure response
iv. SIP Call Setup
1. A direct call can be set up if a user agent has knowledge of how to get to
the destination endpoint, or if it can use local tools to look up the
destination IP address
2. Direct Connection Sequence
a. UAC sends an INVITE messages to the UAS
b. Call parameters are agreed on
c. UAC sends and ACK to the UAS
3. Call Setup with a Proxy Server
a. UAC sends an INVITE message to the proxy server
b. Proxy server resolves the destination address and forwards the
INVITE message to the UAS on behalf of the UAC
4. Call Setup with a Redirect Server
a. UAC sends an INVITE message to the redirect server
b. Redirect server resolves the destination address and responds to
the UAC with a 3XX Redirect message, which contains the
destination address that the UAC can use to send a subsequent
INVITE message
v. SIP Redundancy
1. Install multiple proxy or redirect servers
2. Configure user agent with multiple proxy and/or redirect server entries
vi. SIP Configuration
1. SIP Gateway Syntax
a. sip-ua – enables the SIP user agent on the gateway and enters
SIP user agent configuration mode
b. sip-server dns:SIP_Server_DNS_Name – specifies the DNS
name of a SIP server (e.g. a proxy server) and is issued from SIP
user agent configuration mode
c. session protocol sipv2 – enables a dial peer to use the SIP
protocol and is issued from dial peer configuration mode
d. session target sip-server – issued in dial peer configuration
mode, this command tells a dial peer to point to a SIP server
defined in by the sip-server command which was defined in SIP
user agent configuration mode
2. Example
a. sip-ua
sip-server dns:s1.domain.com

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!
dial-peer voice 100 voip
destination-pattern 5…
session protocol sipv2
session target sip-server
3. Verification and Troubleshooting Commands
a. show sip-ua retry
b. show sip-ua statistics
c. show sip-ua status
d. show sip-ua timers
e. debug voip ccapi inout
c. MGCP
i. MGCP Characteristics
1. Client-server protocol
2. Developed by Cisco, but now an IETF standard
3. Sends messages in plain text
ii. MGCP Components – physical pieces of an MGCP network
1. Endpoint
a. the interface between the VoIP network and the traditional
telephony network (e.g. an FXS port in a Cisco router acting as
an MGCP gateway)
b. Example of an MGCP endpoint identifier: AALN/S1/SU0/0@R1
2. Gateway
a. Converts audio between the VoIP network and the switched
circuit network
b. Examples of MGCP gateways
i. ISUP (ISDN User Part) – supports digital circuit endpoints
using Q.931
ii. NAS (Network Access Server) – supports connections
with modem endpoints
iii. Access Gateway – supports analog and digital endpoints
connected to a PBX
iv. Residential Gateway – supports traditional analog
telephony interfaces
3. Call Agent
a. Controls the gateways and their endpoints
b. The call agent in a Cisco environment is Cisco Unified
CallManager
iii. MGCP Concepts – logical pieces of an MGCP network
1. Calls and Connections
a. Call agent acts as an intermediary point for setting up a call or a
connection
b. Multipoint Call – call agent can instruct an endpoint to
communicate with multiple endpoints
2. Events and Signals – call agent
i. Endpoint reports an “event” to the call agent
ii. Call agent tells the endpoint to send a specific signal in
response to the event (e.g. in the “event” of a phone
going off-hook, play the “signal” of dial-tone)
3. Packages and Digit Maps

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a. Package – a logical grouping of events and signals that an MGCP

gateway can support
b. Digit Map – a dial plan downloaded from the call agent, which
allows an MGCP gateway to match a dial string, as opposed to
the call agent matching the dial string
iv. MGCP Messages
1. Endpoint Configuration (EPCF)
2. Notification Request (RQNT)
3. Notify (NTFY)
4. Create Connection (CRCX)
5. Modify Connection (MDCX)
6. Delete Connection (DLCX)
7. Audit Endpoint (AUEP)
8. Audit Connection (AUCX)
9. Restart In Progress (RSIP)
v. MGCP Call Setup Example
1. Call agent sends RQNT messages to each gateway telling the gateways
to watch for an off-hook event and to supply a dial-tone signal if that event
occurs
2. Phone goes off-hook and receives a dial-tone
3. Gateway collects digits from an attached analog phone, since the
gateway has a digit map, which was downloaded from the call agent
4. Originating gateway notifies the call agent that an off-hook event occurred
and provides the call agent with the dialed digits
5. Call agent tells the originating gateway to create a connection with its
endpoint
6. Originating gateway responds with a session description, including IP
address and UDP port to use for the RTP connection
7. Call agent sends a CRCX to destination gateway, along with the session
information, and the destination gateway begins sending ringing voltage
to an attached analog phone until an off-hook event occurs
8. Destination gateway responds with its session description
9. Call agent forwards that information to the originating gateway
10. RTP stream begins between the two gateways, since the gateways both
know the session descriptions
11. One analog phone goes on-hook, causing its local gateway to send
information of the on-hook event to the call agent, which sends a DLCX
message to each gateway
vi. MGCP Redundancy
1. MGCP Switchover and Switchback
a. Uses two or more call agents, and when a gateway doesn’t see
any MGCP messages from the call agent for a period of time, the
gateway sends keepalive packets, and if no response is received,
the gateway attempts to establish a connection with a backup call
agent
b. Gateway can be configured to switchback to the primary call
agent if it becomes available
2. MGCP Gateway Fallback
a. Works with SRST to maintain a remote office that is connected to
a centralized call agent

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b. If WAN connection which connects the gateway back to the call

agent fails, the gateway continues to operate as an H.323
gateway
vii. MGCP Configuration
1. MGCP Gateway Syntax
a. mgcp – enables MGCP
b. mgcp call-agent IP_Address – specifies the IP address of the
call agent to which the MGCP gateway registers
c. ccm-manager mgcp – allows an MGCP gateway to
communicate with a Cisco Unified CallManager server
d. application MGCPAPP – issued in dial peer configuration mode,
configures a dial peer to run the MGCP application
2. Example
a. ccm-manager mgcp
!
mgcp
mgcp call-agent 192.168.0.25
!
dial-peer voice 100 pots
application MGCPAPP
port 1/0/1
3. Verification and Troubleshooting Commands
a. show mgcp
b. show mgcp connection
c. show mgcp endpoint
d. show mgcp statistics
e. debug mgcp
VII. Quality of Service
a. Design Considerations for Voice Quality
i. Audio Clarity Factors
1. Fidelity – the frequency range of the original signal that is maintained
2. Echo – a reflection of a signal typically caused by an impedance
mismatch
3. Jitter – the uneven arrival of packets
4. Delay – the time for speech to travel across the VoIP network and be
heard at the far end
5. Sidetone – the ability to hear your own voice in the receiver, in order to
make the conversation feel more natural
6. Background Noise – white noise superimposed on the VoIP call to
prevent periods of silence from being interpreted as a disconnected call
ii. Types of Delay
1. Fixed Delay
a. Delay that does not vary during a conversation
b. Examples of fixed delay
i. Coder delay
ii. Serialization delay
iii. Propagation delay
2. Variable Delay
a. Delay that does vary during a conversation
b. For example, queuing delay

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iii. Acceptable Delay

1. Defined by the ITU-T G.114 recommendation
2. 0 – 150 ms of delay – acceptable for most users
3. 150 – 400 ms of delay – might be acceptable if administrators and users
are aware of the impact of delay
4. Above 400 ms – generally an unacceptable level of delay
iv. Packet Drops
1. A Cisco VoIP packet contains by default, 20 ms of voice
2. DSPs can correct for approximately 20 ms to 50 ms of lost voice
v. Quantifying Voice Quality
1. Mean Opinion Score (MOS)
a. Measures voice quality on a scale of 1 through 5, where 1 is
worst, and 5 is best
b. Toll quality voice has an MOS score of approximately 4.0
c. Uses a “trained ear” to judge audio quality
d. G.711 codec has MOS of 4.10
e. G.729 codec has MOS of 3.92
f. G.729a codec has MIS of 3.90
2. Perceptual Speech Quality Measurement (PSQM)
a. Measures voice quality on a scale of 0 through 6.5, where 0 is
best, and 6.5 is worst
b. Uses circuitry, as opposed to a trained ear, to measure audio
quality
3. Perceptual Evaluation of Speech Quality (PESQ)
a. Uses circuitry, as opposed to a trained ear, to measure audio
quality
b. Attempts to match MOS scores
c. Factors in the effects of jitter and packet loss
vi. Goals of QoS
1. Provide dedicated bandwidth to application classes
2. Reduce packet loss, especially for more important application classes
3. Queuing (i.e. congestion management)
4. Congestion avoidance
5. Shaping network traffic (i.e. limiting the bandwidth used by specific
application classes)
6. Prioritizing application classes
b. AutoQoS
i. AutoQoS Features
1. Application classification (i.e. using access lists and NBAR)
2. Policy generation (e.g. Low Latency Queuing and RTP header
compression)
3. Configuration (e.g. using the three-step MQC (Modular Quality of Service
Command Line Interface) approach)
4. Monitoring and reporting (e.g. RMON)
5. Consistency (i.e. same policy applied to multiple devices)
ii. Syntax
1. auto qos voip
2. Can be issued in either interface configuration mode or Frame Relay
DLCI configuration mode
c. Call Admission Control (CAC)

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i. Prevents too many simultaneous voice calls from oversubscribing the available
WAN bandwidth
ii. Categories of CAC
1. Local-based CAC – makes admission decisions based on local router
information (e.g. if a router’s serial interface is up or down)
2. Measurement-based CAC – sends probes out into the network to
measure how the network treats the probes (e.g. in terms of jitter and/or
packet loss)
3. Resource-based CAC – allows network devices, such as routers and/or
gatekeepers, to permit or deny a call based on available resources (e.g.
network bandwidth)
iii. CAC Mechanisms
1. H.323 CAC
a. Resource Availability Indicator (RAI)
i. Configures two resource utilization thresholds (i.e. low
and high)
ii. Calls are rejected after high threshold is exceeded
iii. Calls are once again permitted after resource utilization
drops back below the low threshold
iv. call threshold {global trigger-name | interface
interface_identifier int-calls} low value high value
[busyout | treatment]
b. Call Spike
i. Sets a limit on the maximum number of incoming calls
during a specified time
ii. call spike call-number [steps number-of-steps size
milliseconds]
c. Call Treatment
i. Specifies how rejected calls are treated (e.g. supplying a
specific reason for the rejection using an ISDN cause
code)
ii. call treatment {on | action action [value] | cause-code
causecode | isdn-reject value}
2. SIP CAC
a. Service Assurance Agent (SAA) Response Time Reporter (RTR)
Responder
i. Configures a destination node to respond to a received
SAA probe
ii. rtr responder
b. PSTN Fallback
i. Performs CAC on a call-by-call basis, and grants
admission for the call if the SAA probe was treated
satisfactorily by the network
ii. call fallback active
c. Resource Availability Check
i. Performs CAC based on specified levels of resource
utilization (e.g. memory and CPU utilization)
ii. call threshold global trigger-name low value high
value [busyout][treatment]

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iii. call treatment {on | action action [value]| cause-code

causecode | isdn-reject value}
iv. call threshold interface interface-name interface-
number int-calls low value high value
3. MGCP CAC
a. System Resource Check (SRC) CAC
i. Performs CAC based on available memory and
resources in a gateway
ii. call threshold global trigger-name low value high
value treatment
b. Resource Reservation Protocol (RSVP) CAC
i. Performs CAC using RSVP, which signals the routers
along the call path and requests a bandwidth reservation
for the duration of the call
ii. ip rsvp bandwidth (interface-kbps [single-flow-kbps])
c. Cisco Service Assurance Agent (SAA) CAC
i. Performs CAC based on SAA probes sent out into the
network
ii. call fallback active
iii. mgcp rtrcac
iv. rtr responder
4. Unified CallManager CAC
a. Locations CAC – used for centralized deployments
b. Gatekeeper Zone CAC – used for distributed deployments

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