Introduction to Voice

Gateways

Understanding Cisco Unified

Communications Networks and the
Role of Gateways

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-1

 Outline

 Cisco Unified Communications

 Cisco Unified Communications Gateways
 Gateways in Cisco Unified Communications Deployment
Models
 Gateway Hardware Platforms
 Gateway Operational Modes

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-2

 Cisco Unified
Communications

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 Traditional Telephony Network
Edge
Devices

CO CO
Tie Tie
Trunks Trunks

PBX Switch Switch PBX

CO CO
Trunks Trunks
Local Local
Loops Loops
San Jose Boston
PSTN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-4

 Cisco Unified Communications
Overview

 Integrated solution
 Includes voice, video, data, and mobile applications
 Builds on Cisco Borderless Networks as a secure network
architecture for all communications

IP Mobile Customer Enterprise

Communications Telepresence Conferencing Messaging Social Software
Applications Care

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-5

 Cisco Unified Communications Architecture

Endpoints 5
Cisco Unified IP Wireless Unified IP Unified Personal IP Mobile
Phones IP Phones Phone 7985 Communicator Communicator Phones

Applications
Cisco Unity Unified MeetingPlace Unified Customer Unified Video Unified Personal IP Mobile
Messaging Conferencing Contact Advantage Communicator Communicator Communicator

Services
Smart Business Unified CM Cisco Unified Unified CM
Communications Sys Express Presence Business Edition Unified CM/SME/IME

Infrastructure
Routing Switching Availability Management QoS Security Administration

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-6

 Cisco Unified Communications
Business Benefits

 Cost savings
 Flexibility
 Advanced features:
– Advanced call routing
– Unified messaging
– Integrated information systems
– Long-distance toll bypass
– Voice security
– Customer relationship
– Telephony application services
– Telepresence
– Conferencing

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-7

 Cisco Unified
Communications
Gateways

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-8

 Gateway Functionality

 Unified communication gateways

connect voice-enabled
communication networks together.
 Specifically, they can fulfill these
tasks:
– Switch voice channels between
connected analog and digital
voice circuits
– Convert voice formats between
traditional and VoIP networks
– Interconnect two logically
separate VoIP networks

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-9

 Gateway Functionality (Cont.)

 Supports these VoIP signaling protocols:

– H.323
– MGCP
– SIP
– SCCP
 Works with redundant Cisco Unified Communications
Managers
 Enables call survivability
 Provides analog/digital interfaces to a PBX and the PSTN
 Provides fax/modem services

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-10

 VoIP Signaling Protocols
H.323: Peer-to-Peer Gateway Architecture
 Each gateway maintains the dial plan.

Gatekeeper
Multipoint ITSP
Control Unit
H.323 PBX
Terminal

Cisco
Unified Border
Element IP WAN

PSTN

H.323 Q.921
Q.931
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-11
 VoIP Signaling Protocols (Cont.)
MGCP: Client Server Architecture
 Cisco Unified Communications Manager maintains the dial
plan.

Residential Gateway
E&M

MGCP FXS
Trunking Gateway

PSTN

MGCP Q.921
Q.931

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-12

 VoIP Signaling Protocols (Cont.)
SIP: Peer-to-Peer Gateway Architecture
 Each gateway maintains the dial plan.
 IETF RFC, ASCII-text-based, WWW logic.

ITSP

Cisco
SIP Unified Border
Element IP WAN

PSTN
Q.921
SIP Q.931

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-13

 VoIP Signaling Protocols (Cont.)
SCCP: Client Server Architecture
 Cisco Unified Communications Manager servers maintain the
dial plan.

SCCP

SCCP FXS

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-14

 Gateway Deployment Example

Headquarters Branch
IP WAN

MGCP/ H.323/SIP
H.323/SIP Gateway
Gateway
PSTN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-15

 Gateways in Cisco
Unified Communications
Deployment Models

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 Cisco Unified Communications
Deployment Models
Gateways support these IP telephony deployment models:
 Single-site deployment
 Multisite WAN with centralized call processing
 Multisite WAN with distributed call processing
 Clustering over the IP WAN
Applications

Applications
Cisco Cisco
Unified CM Unified CM
Cluster Cluster

PSTN

IP WAN Branch

Headquarters

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-17

 Single-Site Deployment
Overview
 Cisco Unified Communications
Manager servers, applications, Cisco Unified
Communications
and DSP resources at the same Manager Cluster
physical location
 IP WAN used for data traffic only
 PSTN used for all external calls

PSTN
ta
Da ly SIP or SCCP
On

WAN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-18

 Single-Site Deployment (Cont.)
Gateway Functions
Cisco Unified Communications gateway:
 Uses H.323 or MGCP for PSTN gateways
– Maintains the dial plan with H.323 or SIP
– Receives instructions from the MGCP call agent
 Uses a single best-quality codec for all endpoints (G.711).
 Provides DSP resources for conferencing and media
termination
 Offers appropriate services:
– HSRP for gateway high availability
– QoS mechanisms
– Security

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-19

 Multisite WAN with Centralized Call
Processing
Overview
 Cisco Unified Communications
Manager at central site;
applications centralized or
distributed
 IP WAN carries voice traffic
and call control signaling SIP or SCCP

 Call Admission Control

(limit number of calls per site) IP
PSTN WAN
 SRST for remote branches SRST-
SRST-
Capable
 AAR used if WAN bandwidth is Capable
exceeded

SIP or SCCP SIP or SCCP

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-20
 Multisite WAN with Centralized Call
Processing (Cont.)
Gateway Functions
Cisco Unified Communications gateway:
 Uses H.323 or MGCP for PSTN gateways
 Applies QoS to minimize delay between
Cisco Unified Communications Manager and remote
locations to reduce voice cut-through delays
 At the remote sites, uses SRST, Cisco
Unified Communications Manager Express in SRST mode,
SIP SRST, and MGCP gateway fallback to ensure call-
processing survivability in the event of a WAN failure
 Runs HSRP for redundancy and high availability
 Provides DSP resources for conferencing and media
termination

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-21

 Multisite WAN with Distributed Call
Processing
Overview
 Cisco Unified Cisco Unified
Communications
Communications Manager Cluster
Manager and
applications SIP or SCCP
located at each GK Gatekeeper
site
PSTN IP
 IP WAN carries WAN
intercluster call
control signaling
SIP or SCCP
 Scales to
hundreds of sites
SIP or SCCP
 Transparent use
of the PSTN if the
IP WAN is
Cisco Unified
unavailable
Communications
Manager Cluster
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-22
 Multisite WAN with Distributed Call
Processing (Cont.)
Gateway Functions
Cisco Unified Communications gateway:
 Uses H.323 or MGCP for PSTN gateways
 Participates in H.323 gatekeeper-based CAC
 Uses a single low-bandwidth WAN codec to save bandwidth
and minimize transcoding
 Provides DSP resources for transcoding, conferencing, and
media termination
 Applies QoS for low latency in the IP WAN to ensure timely
VoIP forwarding
 Runs HSRP for redundancy and high availability

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-23

 Clustering over the IP WAN
Overview
 Applications and Cisco Unified Communications Manager
systems of the same cluster distributed over the IP WAN
 IP WAN carries intracluster communication in addition to call
signaling and media
 Call Admission Control (limit number of calls per site)
 AAR used if WAN bandwidth is exceeded

Publisher PSTN

IP WAN

<80 ms Round-Trip Delay

SIP or SCCP
SIP or SCCP
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-24
 Clustering over the IP WAN (Cont.)
Gateway Functions
Cisco Unified Communications gateway:
 Uses H.323 or MGCP for PSTN gateways
 Applies QoS for low latency in the IP WAN:
– 80-ms maximum RTT for ICCS traffic between any two Cisco
Unified Communications Manager servers in the cluster.
– The ICCS traffic types are classified as either priority or best-
effort. Priority ICCS traffic is marked with IP Precedence 3
(DSCP 24 or PHB CS3). Best-effort ICCS traffic is marked
with IP Precedence 0 (DSCP 0 or PHB BE).
– Expedited forwarding of VoIP packets.
 Provides DSP resources for conferencing and media
termination.
 Runs HSRP for redundancy and high availability.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-25

 Gateway Hardware
Platforms

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 Gateway Hardware Platforms
Modern Enterprise Models

Cisco 2900 Series Routers Cisco 3900 Series Routers

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 Gateway Hardware Platforms (Cont.)
Well-Known Older Enterprise Models

Cisco 2800 Series Routers Cisco 3800 Series Routers

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 Gateway Hardware Platforms (Cont.)
Special Voice Gateways

Cisco ATA 186 Cisco VG248 Gateway

Cisco AS5350XM Cisco AS5400XM

Series Gateways Series Gateways Cisco 7200 Series Routers

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-29

 Gateway Operational
Modes

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-30

 Voice Gateway Overview

Three operational modes:

PBX
 Voice switching
– Switches between multiple PSTN
traditional voice networks
 VoIP gateway
– Converts between traditional PSTN IP
telephony and VoIP
 Cisco Unified Border Element
– IP-to-IP gateway
IP IP
– Converts parameters between
multiple VoIP networks

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-31

 Voice Gateway Call Legs

Gateway call
processing:
 Connects incoming call
leg to outgoing call leg Inbound Outbound
 Two major call leg types
are POTS and VoIP Voice
Gateway
 Applies parameters to
both call legs
Incoming Outgoing
– VoIP parameters are Call Leg Call Leg
negotiated.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-32

 Voice-Switching Gateway
Call Legs
Voice-Switching Gateway:
 Signals calls
– Analog signaling
– SS7, ISDN, QSIG PSTN

 Converts between: Voice-Switching

– Signaling types Gateway
– Voice format (analog,
digital) Incoming Outgoing
Call Leg: Call Leg:
– Interface types (T1/E1,
POTS POTS
FXO, FXS, E&M)
 Uses plain old telephone
service (POTS) call legs

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-33

 VoIP Gateway
Call Legs

Source Destination
R1 R2

POTS IP POTS

Originating Terminating
Gateway Gateway

Call Leg 1 Call Leg 2 Call Leg 3 Call Leg 4

(POTS) (VoIP) (VoIP) (POTS)

R1 Inbound R1 Outbound R2 Inbound R2 Outbound

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-34

 Cisco Unified Border Element
Call Legs
Cisco Unified Border
Element:
 Proxies signaling
– May use different IP IP
signaling on both
sides Cisco Unified
 Proxies or passes media Border Element
– May hide addresses
of media channels Incoming Outgoing
 Uses VoIP call legs Call Leg: Call Leg:
VoIP VoIP
– Negotiates VoIP
parameters

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-35

 Summary

 Cisco Unified Communications architecture integrates IP

communications, mobile applications, customer care,
telepresence, conferencing, and messaging.
 Cisco Unified Communications gateways connect voice-
enabled communication networks.
 Gateways are deployed in one of four modes: single-site,
multisite with centralized processing, multisite with distributed
processing, or clustering over the WAN.
 The newest family of enterprise gateways, Cisco 2900 and
3900 Series Integrated Services Routers, offers rich unified
communications features.
 The incoming and outgoing call leg describes the input and
output procedure for a call processed by the voice gateway.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-36

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-37
 Introduction to Voice
Gateways

Examining Gateway Call Routing and

Call Legs

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-38

 Outline

 Gateway Call Routing Components

 End-to-End Call Routing
 Configuring POTS Dial Peers
 Dial Peer Matching
 Matching Inbound Dial Peers
 Matching Outbound Dial Peers
 Default Dial Peer
 Direct Inward Dialing

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-39

 Gateway Call Routing
Components

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 Inbound and Outbound Dial Peers

 A dial peer is an addressable call endpoint.

 Dial peers establish logical connections between call legs to
complete an end-to-end call.
 A gateway uses two dial peers for each call:
– Inbound: matches the incoming call
– Outbound: matches the call destination

Call Setup Message

Inbound Dial Peer

Call Setup Message

Outbound Dial Peer

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-41

 Most Prevalent Dial-Peer Types

Type of Dial Peer Network Technology

Plain old telephone Maps a dial string to a specific voice port on the local
service (POTS) gateway. The voice port connects the gateway to the
PSTN, PBX, or analog telephone.
VoIP Points to the IP address or DNS name of the destination
VoIP device that terminates the call. This mapping applies
to VoIP protocols such as H.323 or SIP.
Multimedia Mail The dial peer is mapped to the email address of the
over IP (MMoIP) SMTP server. This type of dial peer is used for
store-and-forward fax (on-ramp and off-ramp faxing).

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-42

 POTS and VoIP Dial Peers

Voice
Gateway

Telephone
1001
POTS
port 1/0/0
1/0/0 destination-pattern 1001 Telephone
2001
Voice
Gateway
VoIP
session target ipv4:172.16.1.1
IP
destination-pattern 2001
172.16.1.1

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 VoIP Dial Peer

 Points to either H.323 or SIP device

– MGCP configuration does not use voice network dial
peers
 Sets the attributes of the network connection, such as:
– VoIP codec
– Capability to use voice activity detection (VAD)
– Capability for dual tone multifrequency (DTMF) relay

H.323
VoIP

SIP
VoIP

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-44

 VoIP Dial Peer Examples

Cisco Unified Communications Voice Gateway

Manager Cluster

SIP Proxy

VoIP
VoIP
VoIP
VoIP Voice-Mail Server
VoIP

H.323 Gatekeeper
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-45
 End-to-End Call Routing

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-46

 IP and Call Routing Comparison
IP routing Call routing
Static or dynamic Only static.
IP routing table Dial plan.
IP route Dial peer.
Hop-by-hop routing— Inbound and outbound call legs. The gateway
each router makes an negotiates VoIP parameters with preceding and
independent decision next gateways before a call is forwarded.
Destination-based routing Called number, matched by destination pattern, is
one of many selection criteria.
Longest-match rule The longest-match rule for destination pattern
exists but other criteria have higher priority.
Equal paths Preference can be applied to equal dial peers. If all
criteria are the same, random selection.
Default route Possible. Often points at external gateway or
gatekeeper.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-47

 Call Routing
Multiple Paths
Primary path
R2
10.1.1.2
R1 site code
10.1.1.1 300-555 2001
site code
Dial 2001 200-555 IP WAN 1/0/0

1/0/1 2002

1/0/0
1001 1/1/0 2/1/0 2003
1/1/0
PSTN

Secondary path, call forwarded to 300 555-2001

(requires digit manipulation for routing through PSTN)

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-48

 Call Legs
Source Gateway Perspective

Inbound Outbound Call Leg

Call Leg (VoIP Dial Peer in Primary Path,
(POTS Dial POTS Dial Peer in Secondary Path)
Peer)
2001
Dial 2001 R1: 10.1.1.1 IP WAN 1/0/0

1/0/1 2002
R2: 10.1.1.2
1/0/0
1001 1/1/0 2/1/0 2003
1/1/0
PSTN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-49

 Call Legs (Cont.)
Destination Gateway Perspective

Inbound Call Leg Outbound

(VoIP Dial Peer in Primary Path, Call Leg
POTS Dial Peer in Secondary Path) (POTS Dial
Peer)
2001
Dial 2001 R1: 10.1.1.1 IP WAN 1/0/0

1/0/1 2002
R2: 10.1.1.2
1/0/0
1001 1/1/0 2/1/0 2003
1/1/0
PSTN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-50

 Configuring POTS Dial
Peers

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 Configuring POTS Dial Peers
Call Routing Through PSTN

2001
Dial 2001 R1 1/0/0
R2
1/0/1 2002

1/1/0 PSTN 2/1/0

1/0/0
1001 2003
1/1/0

PSTN Path
(requires digit manipulation for routing through PSTN)

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-52

 Configuring POTS Dial Peers (Cont.)
Unidirectional Call Routing

R1(config)# dial-peer voice 1 pots

R1(config-dialpeer)# destination-pattern
2001
R1(config-dialpeer)# forward-digits all
R1(config-dialpeer)# port 1/1/0

Dial 2001
R2
1/0/1 2001

1/0/0 1/1/0 PSTN 2/1/0

1001 R1

R2(config)# dial-peer voice 1 pots

R2(config-dialpeer)# destination-pattern
2001
R2(config-dialpeer)# port 1/0/1

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-53

 Configuring POTS Dial Peers (Cont.)
Bidirectional Call Routing
R1 R2
1/0/1 2001

1/1/0 PSTN 2/1/0

1/0/0
1001

dial-peer voice 1 pots dial-peer voice 1 pots

destination-pattern 2001 destination-pattern 1001
forward-digits all forward-digits all
port 1/1/0 port 2/1/0
dial-peer voice 2 pots dial-peer voice 2 pots
destination-pattern 1001 destination-pattern 2001
port 1/0/0 port 1/0/1

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-54

 Dial Peer Matching

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 Use of String Matching

 Called number: Dialed Number Identification Service (DNIS)

 Calling number: Automatic Number Identification (ANI)
router(config-dialpeer) #
destination-pattern string
 Matches called number in outbound dial-peer
 Matches calling number in inbound dial-peer

router(config-dialpeer) #
incoming called-number string
 Matches called number in inbound dial-peer

router(config-dialpeer) #
answer-address string
 Matches calling number in inbound dial-peer
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-56
 String-Matching Characters
0-9, A-D,*,# Standard characters are digits 0–9, letters A–D, the asterisk (*), and
the pound sign (#) that appear on dial pads.
Plus sign (+) As first character, indicates E.164 standard number; otherwise,
specifies that the preceding digit occurred one or more times
Period (.) Matches any entered digit (used as a wildcard)
Percent sign (%) Indicates that the preceding digit occurred zero or more times
Question mark (?) Indicates that the preceding digit occurred either zero or one time
Press Ctrl-v to disable context-sensitive help and enter ? character
Circumflex (^) Indicates a match to the beginning of the string
Dollar sign ($) Matches the null string at the end of the string
T Timer character. Indicates a variable-length dial string. Makes the
router wait until all digits are received before routing call
Backslash (\) Followed by a single character, matches that character
Brackets [ ] Indicates a range
Parentheses ( ) Indicates a pattern

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-57

 Number-Matching Examples

Number String Matching telephone numbers

5551234 Matches single number 5551234
^5551234$ Matches single number 5551234
555123[5-9] Matches the numbers 5551235-5551239
55512[3-4]. Matches 7-digit numbers where the first 5 digits are 55512, the
sixth digit is 3 or 4, and the last digit is any digit
.T Matches any number with the length of 1 to 32 digits
(200)?5551234 Matches the numbers 2005551234 and 5551234
1[2-3]%4 Matches numbers that start with 1, have any number of
occurrences of the digit 2 or 3, and end with 4

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-58

 Matching Inbound Dial
Peers

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-59

 Matching Inbound Dial Peers
Elements in the Call Setup Message
To match inbound call legs to dial peers, the gateway
uses one of three elements (ISDN example):
 Called number (DNIS)
– Derived from the ISDN setup message or channel
associated signaling (CAS) DNIS
 Calling number (ANI)
– Derived from the ISDN setup message or CAS ANI
 Inbound voice port

1/0/0
Call Setup Message ANI DNIS POTS

POTS

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-60

 Matching Inbound Dial Peers (Cont.)
Relevant Dial-Peer Attributes
Precedence of matching criteria for inbound dial peers:
1. incoming called-number: Matches called number
– Most explicit match
2. answer-address: Matches calling number
– Most explicit match
3. destination-pattern: Matches calling number
– Most explicit match
4. port: Matches the dial peer with the inbound voice port (POTS only)
– If multiple dial peers have the same port, selects the dial peer
added to the configuration earlier
5. default dial peer: Predefined parameters
 Only one condition must be met.
 The gateway stops searching when a dial-peer match is found.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-61

 Matching Inbound Dial Peers (Cont.)
Example
2001
1/0/0
Dial 2001 R1 R2
1/0/1 2002

1/1/0 PSTN 2/1/0

1/0/0
1001 2003
1/1/0

dial-peer voice 1 pots

destination-pattern 2001
dial-peer voice 1 pots port 1/0/0
destination-pattern 200. dial-peer voice 2 pots
forward-digits all answer-address 100.
port 1/1/0 port 2/1/0
dial-peer voice 3 pots
incoming called-number 100.
Which inbound dial peer port 2/1/0
is selected? dial-peer voice 4 pots
(answer-address) destination-pattern 100.
forward-digits all
port 2/1/0
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-62
 Matching Inbound Dial Peers (Cont.)
Guidelines
 answer-address
– Useful for matching the geographical region of caller
 Callers from a given country directed at the appropriate
language-speaking team
 Callers from a specific region directed at the regional
sales staff
 incoming called-number
– Recommended for most configurations
– Useful for service selection
 Different numbers for sales and technical support
 Different numbers for shipping order, tracking, and
cancellation

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-63

 Matching Outbound Dial
Peers

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-64

 Matching Outbound Dial Peers

Criteria required for outbound dial peers:

 destination-pattern
– Uses the called number to match the outbound dial peer
– Most explicit match rule applies

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-65

 Matching Outbound Dial Peers (Cont.)
Example 2001
1/0/0
R1 R2
User dials
1/0/1 2002

1/1/0 PSTN 2/1/0

1/0/0
1001 2011
1/1/0
dial-peer voice 1 pots
destination-pattern .T 2111
port 1/1/0 1/1/1
dial-peer voice 2 pots
destination-pattern 20[0-1].
forward-digits all
port 1/1/0
dial-peer voice 3 pots Dialed number 2001 matches dial peer 4.
destination-pattern 200. Dialed number 2002 matches dial peer 3.
forward-digits all Dialed number 2011 matches dial peer 2.
port 1/1/0 Dialed number 2111 matches dial peer 1.
dial-peer voice 4 pots
destination-pattern 2001
forward-digits all
port 1/1/0
© 2010 Cisco Systems, Inc. All rights reserved.
? CVOICE v8.0—1-66
 Default Dial Peer

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 Default Dial Peer
No explicit match for inbound dial peer configured

dial-peer voice 1 pots

destination-pattern
200.
forward-digits all
port 1/1/0 2001
Dial 2001 1/0/0
R2
1/0/1 2002

1/0/0 1/1/0 PSTN 2/1/0

1001 2003
R1 1/1/0

What are the inbound dial peers dial-peer voice 1 pots

destination-pattern 2001
when extension 1001 calls 2001?
port 1/0/0

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-68

 Default Dial Peer (Cont.)
Features
When no explicit inbound dial peer is configured, the
gateway uses the default dial peer:
 Also called dial peer 0
 Fails to negotiate any nondefault parameters

Inbound VoIP dial peer 0 Inbound POTS dial peer 0

parameters parameters
G.729 or G.711 codec No applications
No DTMF relay No DID
IP precedence 0
VAD enabled
No RSVP support
Fax-rate voice

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-69

 Default Dial Peer (Cont.)
Guidelines
 Avoid using dial peer 0.
– Calls with nondefault parameters will fail.
– Incoming called-number ensures a match with the
desired parameters.
– Many errors are due to codec, VAD, and DTMF-relay
misconfigurations when dial peer 0 is matched.
 Cisco AS5350, AS5400, and AS5850 Universal Gateways
require explicit inbound dial peers matching:
– To accept incoming POTS calls as voice calls.
– If there is no inbound dial peer match, the call is treated
and processed as a dialup (modem) call.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-70

 Direct Inward Dialing

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-71

 Direct Inward Dialing
Two-Stage Dialing and One-Stage Dialing
 Two-stage dialing
– When a call arrives on a POTS voice port
 POTS voice port is seized inbound
 Gateway presents dial tone and collects digits
 One-stage dialing
– When a call arrives on a DID-enabled voice port
 Gateway does not present a dial tone
 Gateway receives the entire called number
 Enabled through DID on inbound POTS dial peers
 DID not supported on FXS/FXO/E&M analog ports
 DID available on FXS-DID and digital circuits

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-72

 Two-Stage Dialing
Overview
1. User dials 555
2. PSTN delivers call, gateway sends dial tone and starts
collecting remaining digits
3. User hears secondary dial tone and dials 2001
4. Gateway matches the outbound dial peer and signals the call

PSTN
1 3 consumes
555 2 Extensions
Dial 555-2001 555-XXXX
2001
PSTN
1/1/1
1/0/0

dial-peer voice 1 pots

4 destination-pattern 2001
port 1/1/1

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-73

 Two-Stage Dialing (Cont.)
Digit-by-Digit Collection
 Digits are collected in-band
 Outbound dial peer matching done on a digit-by-digit basis
 Gateway matches dial peer after receiving each digit
– Routing when a match is found 2002-2009

PSTN
Dial 555-2001 consumes
Call
555 1/1/0
R2 2001
PSTN
1/1/1
1/0/0

dial-peer voice 1 pots

destination-pattern
2001
port 1/1/1
dial-peer voice 2 pots
destination-pattern 200
port 1/1/0
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-74
 Two-Stage Dialing (Cont.)
Wildcard Use
 Most explicit match rule applies
 Destination gateway matches dial peer 1 for the outbound
call leg
2002-2009

PSTN
consumes
Dial 555-2001 555
R2 1/1/0
2001

PSTN
1/0/0 1/1/1

dial-peer voice 1 pots

destination-pattern 2001
port 1/1/1
dial-peer voice 2 pots
destination-pattern 200.
port 1/1/0

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-75

 Two-Stage Dialing (Cont.)
Gateway-by-Gateway Processing
dial-peer voice 1 pots
2 5 8 3 destination-pattern 55
port 1/0/1
Dial 55 4 2001 …Irregular intervals…

1 … 0 … 0 … 2 … 4 … 5 … 5

…Irregular intervals… 1
1/0/1 R1
4 4 … 2 … 0 … 0 … 1
R2
1/0/1
…Irregular intervals… R3
10
1 … 0 … 0 … 2
1/0/1

dial-peer voice 1 pots 7

6 destination-pattern 4 2001
port 1/0/1 dial-peer voice 1 pots
destination-pattern 2001
9 port 1/0/1
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-76
 Two-Stage Dialing (Cont.)
Variable-Length Numbers
 The gateway waits for remaining digits.
– The default interdigit timeout is 10 seconds.
– The interdigit timeout can be modified using the timeouts
interdigit command in voice-port configuration mode.

dial-peer voice 1 pots

incoming called-number .
!
dial-peer voice 10 pots
destination-pattern .T
port 1/0/0

Dial 555 2001

? … 0 …1sec… 0 …2sec… 2 … 5 … 5 … 5
PSTN
1/0/0

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-77

 One-Stage Dialing
DID Overview
 User dials 555-2001
 PSTN sends the entire called number in one call setup
message to destination gateway (if digital circuit)
 Destination gateway matches the outbound dial peer and
signals the call
Extensions
555-XXXX
1/1/1:0 2
1/0/0:0 2001
1
PSTN
1/1/1

Dial 555-2001 dial-peer voice 1 pots

PSTN incoming called-number .
consumes 3 direct-inward-dial
555
!
dial-peer voice 2 pots
destination-pattern 2001
port 1/1/1

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-78

 One-Stage Dialing (Cont.)
Matching with Complete Called Number
 With DID configured in the inbound POTS dial peer, the
router uses the complete called number to match the
outbound dial peer.
2002-2009

PSTN
consumes
555
Dial 555-2001
1/1/0
2001
PSTN
1/0/0:0 1/1/1

dial-peer voice 1 pots

incoming called-number .
direct-inward-dial
dial-peer voice 2 pots
destination-pattern 2001
port 1/1/1
dial-peer voice 3 pots
destination-pattern 200
port 1/1/0
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-79
 One-Stage Dialing (Cont.)
Gateway-by-Gateway Processing
dial-peer voice 1 pots
2 3 destination-pattern 55.....
port 1/0/1:0
Dial 554-2001 …Irregular intervals…
1 … 0 … 0 … 2 … 4 … 5 … 5

1
1/0/1:0 R1
1/0/0:0 42001
R2
2/0/0:0 R3
6
2001 1/0/1
dial-peer voice 1 pots
4 incoming called-number .
direct-inward-dial dial-peer voice 1 pots
incoming called-number .
2001
dial-peer voice 2 pots
destination-pattern 4.... direct-inward-dial
port 2/0/0:0 5 dial-peer voice 2 pots
destination-pattern 2001
port 1/0/1
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-80
 One-Stage Dialing (Cont.)
Configuring DID

PSTN
1/0/0

dial-peer voice 1 pots

incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
destination-pattern .T
port 1/0/0

String Description
. Matches any number with at least one digit. Useful for outbound
(destination-pattern) and especially inbound matching (incoming called-
number).
.T Matches any number with at least one digit. The timer character matches
either the interdigit timeout or the termination character (#). Useful for
outbound matching (destination-pattern).

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-81

 Summary

 The most common dial peer types are VoIP and POTS.
 In call routing, each gateway identifies the inbound and
outbound dial peer.
 POTS dial peers facilitate calling over POTS ports.
 Telephone numbers are matched using a sequence of standard
and special characters.
 The matching order for inbound dial peer is: incoming dialed-
number, answer-address, destination-pattern, and port.
 The outbound dial peer is found by using the longest match of
the destination-pattern command.
 If no explicit inbound dial peer is identified, the default peer 0 is
used to set the parameters to predefined values.
 DID enables the matching of the entire number instead of digit-
by-digit matching.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-82

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-83
 Introduction to Voice
Gateways

Configuring Gateway Voice Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-84

 Outline

 Voice Ports Overview

 Analog Voice Ports
 Configuring Analog Voice Ports
 Digital Voice Ports
 Understanding ISDN
 Configuring Digital Voice Ports
 Configuring ISDN
 Fine-Tuning Analog and Digital Voice Ports
 Echo Cancellation
 Verifying Analog and Digital Voice Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-85

 Voice Ports Overview

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-86

 Voice Port Overview
Connecting End User Equipment

Voice Port Voice Port

PSTN
FXS T1 or E1 or
(Analog) ISDN (Digital)

Voice Port Voice Port

PSTN
FXS FXO
(Analog) (Analog)

Voice Ports
Voice Port
FXO
(Analog)
PSTN
FXS
(Analog) FXS-DID
(Analog)
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-87
 Voice Port Overview (Cont.)
PBX Trunks
Voice Port Voice Port

PSTN
E&M or FXO FXO (Analog)
(Analog) additional FXS-DID
possible
Voice Port Voice Port

T1, E1,or PSTN

T1, E1,or
ISDN ISDN
(Digital) (Digital)

Voice Port Voice Port Voice Port Voice Port

PSTN
T1, E1, T1, E1, T1, E1, T1, E1,
or ISDN or ISDN or ISDN or ISDN
QSIG (Digital) (Digital) QSIG
(Digital) (Digital)

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-88

 Voice Port Overview (Cont.)
Centralized Automated Message Accounting Trunks

T1 PRI for Standard Calls

PSTN

CAMA Trunk
for Emergency Public Safety
Calls Answering Point

Centralized Automated Message Accounting (CAMA) available in North America only

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-89

 Voice Trunk Example
Chicago T1 PRI

E&M
Trunk
PSTN
San Jose
T1 QSIG
Trunk

T1 CAS*
Trunk
E1 R2 E1 CCS**
London Trunk Rome Trunk

Denver T1 QSIG
Trunk

T1 PRI

*CAS = Channel associated signaling

**CCS = Common channel signaling
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-90
 Installing Voice Ports

Cisco 2921/2951 Series router rear panel

EHWIC EHWIC EHWIC EHWIC

slot 3 slot 2 slot 1 slot 0

Service module Service module

slot 2 (SM2) slot 1 (SM1)

Interface card slot 1 Interface card slot 0

voice-port 2/0/1
no shutdown
Interface 1 Interface 0

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-91

 Analog Voice Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-92

 Analog Voice Ports

FXS

 FXS: Connects directly to end-user equipment such as telephones, fax

machines, or modems

FXO FXO
PSTN

 FXO: Used for trunk, or tie-line, connections to a PSTN CO or to a PBX

that does not support E&M signaling

E&M E&M
WAN or
PSTN

 E&M: Used for trunk circuits to connect telephone switches to each other

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-93

 Analog Signaling Overview

 Supervisory signaling for FXO/FXS

– Loop-start
– Ground-start
 Address signaling
– Pulse
– Dual tone multifrequency
 Informational signaling
– Call progress tones

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-94

 Analog Signaling
Supervisory Signaling Comparison
FXS / FXO E&M
Wiring RJ-11, two voice wires, 8-pin modular, up to 8 wires in
used for both total: 2 or 4 voice wires (half or
supervisory signaling full duplex audio path) + 2 half or
and voice modulation full duplex wires for supervisory
signaling (M-lead, E-lead) + 2 for
voice modulation
Types of Loop-start (more Type I, II, III, IV, V, SSDC5
supervisory common), ground-start
signaling
Supervisory Differences in voltage Differences in voltage and
signaling and grounding on the tip grounding on the M-lead and E-
differences and ring lines lead
Access signaling N/A Immediate-start, wink-start, delay-
start, followed by pulse or DTMF

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-95

 Address Signaling—DTMF

1209 1336 1447

697

770

852

941

Frequency Tone Matrix [Hz]

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-96

 Call Progress Tones
North American call progress tones example:

Tone Frequency (Hz) On Off

Dial 350 + 440 Continuous Continuous
Busy 480 + 620 0.5 0.5
Ringback, normal 440 + 480 2 4
Ringback, PBX 440 + 480 1 3
Congestion (toll) 480 + 620 0.2 0.3

Reorder (local) 480 + 620 0.3 0.2

Receiver off-hook 1400 + 2060 + 0.1 0.1
2450 + 2600
No such number 200–400 Continuous, 1- Continuous, 1-Hz
Hz frequency frequency
modulation modulation

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-97

 Configuring Analog Voice
Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-98

 Configuration Overview

Configuration tasks:
 FXS
– CP tones, signaling (loop-start, ground-start)
 FXO, FXS-DID
– Signaling (loop-start, ground-start)
 E&M (Ear and Mouth)
– Operation (two-wire, four-wire)
– Type (1,2,3,5)
– Signaling (immediate-start, wink-start, delay-start)
 Centralized Automated Message Accounting
– Signaling (CAMA)
– ANI mapping

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-99

 Configuring FXS Voice Ports

Voice Port
0/2/0

WAN

voice-port 0/2/0
signal loopstart
cptone GB
ring cadence pattern01
no shutdown

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-100

 Configuring FXO Voice Ports

FXO emulates end-user equipment connected to the CO switch.

 DID not supported on FXO
– DNIS not received from CO switch
 Incoming calls directed to internal number using private line, automatic
ringdown (PLAR)

voice-port 0/0/0
signal loopstart
connection plar opx 4001
dial-peer voice 90 pots
destination-pattern 9T
port 0/0/0

0/0/0
PSTN
4001 FXO

Inbound calls should

4002 be routed to 4001.
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-101
 Configuring FXS-DID Voice Ports

Inbound calls routed

directly to internal
phones. Dial 501-1001
DID Support 0/0/0 FXS DID Inbound 0/0/0
PSTN
0/1/0 FXO Outbound 0/1/0

501-1001
voice-port 0/0/0
501-1002 signal did wink-start
voice-port 0/1/0
signal groundstart
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 2 pots
destination-pattern 9[2-8].........
port 0/1/0

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-102

 Configuring E&M Voice Ports
E&M Trunk
Wink-Start
Type I 2001
1001 Two-Wire
PBX
2002
1/1/1
1002 2003
Inbound DNIS
Outbound DNIS
1003 2004
voice-port 1/1/1
signal wink-start
operation 2-wire
type 1
no shutdown
exit
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 2 pots
destination-pattern 1...
forward-digits all
port 1/1/1
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-103
 Configuring CAMA Voice Ports

T1 PRI for Standard Calls

0/0/0
PSTN
1/1/1

CAMA Trunk Public Safety

for Emergency Answering
Calls Point

voice-port 1/1/1 dial-peer voice 911 pots

ani mapping 1 312 destination-pattern 911
signal cama KP-NPD-NXX-XXXX-ST port 1/1/1
dial-peer voice 1 pots forward-digits all
incoming called-number . !
direct-inward-dial dial-peer voice 9911 pots
dial-peer voice 90 pots destination-pattern 9911
destination-pattern 9[2-8]......... port 1/1/1
port 0/0/0:23 forward-digits 3

Centralized Automated Message Accounting (CAMA) available in North America only

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-104

 Digital Voice Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-105

 Digital Voice Ports Overview

 T1: Uses time-division multiplexing (TDM) to transmit digital

data over 24 voice channels using channel associated
signaling (CAS)
 E1: Uses TDM to transmit digital data over 32 timeslots,
including 30 voice channels, 1 framing channel, and 1
signaling channel
 ISDN: A circuit-switched telephone network system designed
to allow digital transmission of voice and data over ordinary
telephone copper wires:
– BRI: 128 kb/s; 2 B channels and 1 D channel
– T1 PRI: 1.472 Mb/s; 23 B channels and 1 D channel
– E1 PRI: 1.920 Mb/s; 30 B channels and 1 D channel
– Uses common channel signaling (CCS)

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-106

 Digital Circuit Types

Type Circuit Option Description

Digital T1/E1 CAS  Analog signaling over digital T1/E1
E1 R2 CAS  Can provide ANI calling party ID (caller ID)
ISDN T1 PRI  More services than CAS
(CCS) E1 PRI  Separate signaling channel (D channel)
 Common on modern PBXs
PRI NFAS  Multiple ISDN PRI interfaces controlled by a
single D channel
 Backup D channel can be configured
BRI  Mostly for Europe, Middle East, and Africa
QSIG  Created for interoperation of PBXs from
different vendors
 Rich in supplementary services

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-107

 T1 CAS Overview

 T1 CAS uses in-band robbed-bit signaling.

 Signaling for a particular traffic circuit is permanently
associated with that circuit.
 Signaling is based on analog signaling: loop-start, ground-
start, and E&M variants.
 E&M supports various feature groups.

No D Channel Required
Analog Signaling

24 B Channels (Voice)

T1 CAS

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-108

 T1 CAS Super Frame Format

Time Slot
8 Bits
24 * 8 bits + 1 bit = 1 frame (193 bits)

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24

1 Bit Sync.
12 Frames = Super Frame

1 2 3 4 5 6 7 8 9 10 11 12

24 * (7 bits + 1 robbed bit) + 1 bit = 1 frame (193 bits)

Time Slot
7 Bits +
1 Robbed Bit

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-109

 T1 CAS Extended Super Frame Format

Time Slot
8 Bits
24 * 8 bits + 1 bit = 1 frame (193 bits)

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24

1 Bit
24 Frames = Extended Super Frame Sync.

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24

24 * (7 bits + 1 robbed bit) + 1 bit = 1 frame (193 bits)

Time Slot
7 Bits +
1 Robbed Bit

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-110

 E1 CAS Overview

E1 uses multiframe format:

 Consists of 16 consecutive 256-bit frames
 Consists of 32 time slots:
– One slot for frame synchronization
– One slot for signaling
– 30 slots for actual voice traffic
 E1 R2 variant supports inbound and outbound DNIS and ANI
– Other variants may not support inbound ANI

No D Channel
Analog Signaling
30 B Channels (Voice)
E1 CAS

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-111

 E1 CAS Multiframe Format

Time Slot 1 Time slot 17

Frame synchronization Frame 1 Indicates start of multiframe
Frames 2–16 Carry signaling (ABCD bits) for two voice channels

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32
1. Frame: Start of Multiframe
2. Frame: Signaling for Voice Slots 2 and 18
3. Frame: Signaling for Voice Slots 3 and 19
4. Frame: Signaling for Voice Slots 4 and 20
5. Frame: Signaling for Voice Slots 5 and 21
16 Frames 6. Frame: Signaling for Voice Slots 6 and 22
= 7. Frame: Signaling for Voice Slots 7 and 23
Multiframe 8. Frame: Signaling for Voice Slots 8 and 24
2.048 Mb/s 9. Frame: Signaling for Voice Slots 9 and 25
10. Frame: Signaling for Voice Slots 10 and 26
11. Frame: Signaling for Voice Slots 11 and 27
12. Frame: Signaling for Voice Slots 12 and 28
13. Frame: Signaling for Voice Slots 13 and 29
14. Frame: Signaling for Voice Slots 14 and 30
15. Frame: Signaling for Voice Slots 15 and 31
16. Frame: Signaling for Voice Slots 16 and 32

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-112

 ISDN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-113

 ISDN
Common Channel Signaling (CCS)
 Voice, video, and data are sent over separate channels.

D Channel 16 kb/s (Signaling)

ISDN BRI 2 B channels (Voice)

D Channel 64 kb/s (Signaling)

ISDN T1 PRI 23 B Channels (Voice)

D Channel 64 kb/s (Signaling)

ISDN E1 PRI 30 B Channels (Voice)

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-114

 ISDN BRI and PRI Interfaces

BRI T1 PRI E1 PRI

B Channels 2 x 64 kb/s 23 x 64 kb/s 30 x 64 kb/s
D Channels 1 x 16 kb/s 1 x 64 kb/s 1 x 64 kb/s
Framing Rates 16 kb/s 8 kb/s 64 kb/s
Frame Formats NT, TE frame SF, ESF Multiframe (CRC-4)
Line Coding 2B1Q or 4B3T AMI or B8ZS HDB3
Country World North America, Japan Europe, Australia

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-115

 ISDN Architecture

Layer 2 Layer 3
Protocol Q.921 Q.931
name
Features Link Access Procedure on the D Various message types:
channel (LAPD). Similar to HDLC. - call-establishment
Provides error detection and
correction. - call-termination
- information
- miscellaneous
Task Provide terminal endpoint identifiers Supports these connections:
(TEIs) as Layer 2 addresses to end - user-to-user
devices:
- circuit-switched
- through static configuration
- packet-switched
- dynamically allocated by PSTN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-116

 Non-Facility Associated Signaling

 Allows a single D channel to control multiple PRI interfaces.

 A backup D channel can be configured, but only the NFAS
primary D channel must be configured.
 NFAS is only supported with a channelized T1 controller.

D Channel 64 kb/s (Signaling)

ISDN T1 PRI 23 B Channels (Voice)

D Channel 64 kb/s (Signaling)

ISDN T1 PRI NFAS 23 B Channels (Voice)

24 B Channels (Voice)

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-117

 Configuring Digital Voice
Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-118

 Configuring Digital Voice Ports
T1/E1 Voice Port Configuration Overview
Configure controller settings:
 Framing
 Line encoding
 Clock source
 Create digital voice ports:
– DS0 group
– Time slots
– Signal type

Configure voice port parameters:

 compand-type
 cptone

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-119

 Configuring Digital Ports
DS0 Group Configuration

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-120

 Configuring Digital Ports (Cont.)
Clock Sources
Tx
E1
Clock
1/0

controller e1 1/0
framing crc4 PBX
linecode hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start

Rx
Clock
T1 1/0
PSTN
Tx

controller t1 1/0
framing esf
linecode ami
clock source line
ds0-group timeslots 1-12 type e&m-wink-start

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-121

 Configuring Digital Ports (Cont.)
Logical Voice Port Configuration

T1 CAS
E&M Wink-Start
PSTN

voice-port 1/0:1
cptone US
compand-type u-law
no shutdown

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-122

 Configuring Digital Ports (Cont.)
T1 CAS: Inbound E&M FGD and Outbound FGD-EANA
E&M FGD* Time Slots 1 to 12, Receive ANI

PSTN

E&M FGD EANA** Time Slots 13 to 24, Send ANI

controller T1 0/0/0
framing esf
linecode b8zs
ds0-group 0 timeslots 1-12 type e&m-fgd
ds0-group 1 timeslots 13-24 type fgd-eana
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 90 pots
destination-pattern 9T
port 0/0/0:1

*FGD = E&M Feature Group D; provides inbound and outbound DNIS, and inbound ANI
**EANA = FGD-Exchange Access North American; provides inbound and outbound DNIS, and
outbound ANI
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-123
 Configuring Digital Ports (Cont.)
E1 R2 CAS

E1 R2 CAS
PSTN
Inbound DNIS and ANI
Outbound DNIS and ANI

controller e1 0/0/0
ds0-group 0 timeslots 1-31 type r2-digital r2-compelled ani
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 90 pots
destination-pattern 9T
port 0/0/0:0
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-124
 Configuring ISDN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-125

 Configuring ISDN
Configuration Overview
 BRI:
– Interface configuration:
 isdn switch-type
 isdn incoming-voice
 isdn protocol-emulate
 PRI:
– Global/controller configuration: isdn switch-type
– T1/E1 controller configuration: pri-group
– D-channel (interface) configuration: isdn incoming-voice
– QSIG signaling configuration

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-126

 ISDN BRI Configuration

2 B Channels
+
1 D Channel

ISDN BRI
PSTN
Router1

network-clock-participate wic 0
Phone1-1 Phone1-2 interface bri 0/0
2001 2002 isdn switch-type basic-net3
isdn overlap-receiving
isdn incoming-voice voice
isdn protocol-emulate user

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-127

 ISDN E1 PRI Configuration

30 B Channels
+
1 D Channel

ISDN E1 PRI
PSTN

network-clock-participate wic 0
network-clock-select 1 e1 0/0/0
isdn switch-type primary-net5
controller e1 0/0/0
pri-group timeslots 1-31
interface Serial0/0/0:15
isdn overlap-receiving
isdn incoming-voice voice

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-128

 Fine-Tuning Analog and
Digital Voice Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-129

 Fine-Tuning Voice Ports
Cross-Connect Between Analog Port and Digital DS0
 Cross-connect works for:
– Ports on the same module (NM-HD-2VE)
– CAS DS0 groups consisting of one time slot
– Maximum 4 FXO/FXS ports at the same time

FXO T1/E1 CAS

DS0
PSTN
DS1
FXS

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-130

 Fine-Tuning Voice Ports (Cont.)
Cross-Connect Configuration

e1 0/0
FXO CAS
0/1/0
DS0
PSTN
DS1
FXS
0/1/1

controller e1 0/0
ds0-group 0 timeslots 5 type fxs-ground-start
ds0-group 1 timeslots 8 type e&m-fgd
voice-port 0/0:0
signal loop-start
voice-port 0/0:1
operation 2-wire
type 1
signal wink-start
connect connect1 voice-port 0/1/0 e1 0/0 0
connect connect1 voice-port 0/1/1 e1 0/0 1

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-131

 Fine-Tuning Analog Voice Ports
Timing Parameters
 timeouts initial
– How long the dial tone is presented before the first digit is expected
 timeouts interdigit
– How long to wait for the next digit before the number is considered
complete
 timeouts ringing
– How long a caller may let the telephone ring when there is no answer
 timing digit
– DTMF digit signal duration
 timing interdigit
– DTMF interdigit duration
 timing hookflash-in and hookflash-out
– maximum duration of a hookflash indication
– hookflash is an indication by a caller to do something special with the
call, such as transfer or place the call on hold

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-132

 Fine-Tuning Analog Voice Ports (Cont.)
Timing Parameters Configuration
 Present the dial tone for 15 seconds before the first digit is
collected.
 Keep waiting 15 seconds for the next digit.
 Allow the phone to ring 4 minutes when there is no answer.
 Hookflash indication may not exceed 0.5 seconds.

FXS
0/1/0

voice-port 0/1/0
timeouts initial 15
timeouts interdigit 15
timeouts ringing 240
timing hookflash-in 500

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-133

 Echo Cancellation

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-134

 Echo Origin

 Echo is a signal that leaks from the Rx path into the Tx path.
 Typically, due to impedance mismatch at the two-wire to four-
wire hybrid connections.
 Echo annoys callers if above amplitude and delay threshold.

Four-Wire
Two-Wire Trunk Two-Wire
Subscriber Subscriber
Loop Loop

Hybrid Echo Hybrid

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-135

 Talker Echo

 Signal leaks from Rx to Tx path at the remote end

Talker Echo (Most Common)

Tx Rx

PSTN or
PSTN PBX User
Rx Tx

Hybrid Hybrid

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-136

 Listener Echo

 Signal leaks from Rx to Tx path at both ends

Listener Echo (Less Common)

Tx Rx
PSTN or
PSTN or
PSTN PBX User
PBX User
Rx Tx

Hybrid Hybrid

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-137

 Echo Cancellation
ITU-T G.168 Echo Canceller
 Enabled by default.
 Faces into the PSTN side of a voice gateway.
 Captures and stores the outgoing voice signal.
 Watches the Rx path for echo.
PSTN or
– Estimates the level PSTN
PBX or
User
– Subtracts the original signal from the Rx signal PBX User

 When one side is silent, comfort noise is generated.

 Echo canceller coverage is the length of time that is stored in
memory and can be adjusted.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-138

 Echo Canceller Parameters

 ERL (echo return loss): Represents the reduction of returning echo

(larger is better)
 ERLE (ERL enhancement): Additional echo loss from canceller
 ACOM = ERL + ERLE (larger is better)
 Use output attenuation and input gain to tune ERL to at least 6 dB

Echo Cancel Coverage

(Tail Length) Tail Circuit

Output Attenuation
^
H(t)
ACOM
PSTN or
^y(t) Input Gain PSTN
PBX or
User
SOUT
Sout - SIN
ERL
PBX User

ERLE
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-139
 Configuring Echo Cancellation

router(config-voiceport)#
echo-cancel coverage {8 | 16 | 24 | 32 | 48 | 64}
 Adjusts the time coverage of the echo canceller
 Available options and default value differ per platform and
software version

router(config-voiceport)#
(no) echo-cancel enable
 Disables or re-enables echo canceller
 Default: enabled

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-140

 Verifying Analog and
Digital Voice Ports

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-141

 Verifying Voice Ports
Command Overview

Command Description
show voice port [slot/port | Displays configuration information about a specific
summary] voice port or a summary of all voice ports
show controllers bri slot/port
show controllers t1 slot/port Displays information about the specified voice port
show controllers e1 slot/port
show voice call summary Verifies the call status for all voice ports
show call active voice Displays the contents of the active call table
show call history voice Displays the contents of the call history table

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-142

 show voice port summary Command

Router# show voice port summary

IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:18 19 fxo-ls up dorm idle on-hook y
0:19 20 fxo-ls up dorm idle on-hook y
0:20 21 fxo-ls up dorm idle on-hook y
0:21 22 fxo-ls up dorm idle on-hook y
0:22 23 fxo-ls up dorm idle on-hook y
0:23 24 e&m-imd up dorm idle idle y
1/1 -- fxs-ls up dorm on-hook idle y
1/2 -- fxs-ls up dorm on-hook idle y
1/3 -- e&m-imd up dorm idle idle y
1/4 -- e&m-imd up dorm idle idle y
1/5 -- fxo-ls up dorm idle on-hook y
1/6 -- fxo-ls up dorm idle on-hook y

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-143

 Verifying Analog Voice Ports
show voice port
Router# show voice port
DS0 Group 1:0 - 1:0
Type of VoicePort is CAS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Connection Mode is normal
Connection Number is not set
.
.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-144

 Verifying Voice Ports
show controller
Router# show controller T1 1/0/0
T1 1/0/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
No alarms detected.
alarm-trigger is not set
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
Data in current interval (180 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-145

 Verifying Voice Ports (Cont.)
show voice call summary

Router# show voice call summary

PORT CODEC VAD VTSP STATE VPM STATE

========= ======== === ===================== ==================
1/015.1 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.2 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.3 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.4 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.5 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.6 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.7 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.8 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.9 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.10 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.11 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.12 g729r8 y S_CONNECT S_TSP_CONNECT

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-146

 Verifying Voice Ports (Cont.)
show call active voice
Router# show call active voice
GENERIC:
SetupTime=94523746 ms
Index=448
PeerAddress=##73072

PeerSubAddress=
PeerId=70000

PeerIfIndex=37
LogicalIfIndex=0
ConnectTime=94524043
DisconectTime=94546241
CallOrigin=1

ChargedUnits=0
InfoType=2
TransmitPackets=6251
TransmitBytes=125020
ReceivePackets=3300
ReceiveBytes=66000

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-147

 Verifying Voice Ports (Cont.)
show call history voice
Router# show call history voice
GENERIC:
SetupTime=94893250 ms
Index=450
PeerAddress=##52258
PeerSubAddress=
PeerId=50000
PeerIfIndex=35
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing.

ConnectTime=94893780
DisconectTime=95015500
CallOrigin=1

ChargedUnits=0
InfoType=2
TransmitPackets=32258
TransmitBytes=645160
ReceivePackets=20061
ReceiveBytes=401220
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-148
 Verifying Voice Ports (Cont.)
test Command Overview

Command Description
test voice port slot/subunit/port Tests detector-related functions on a voice
detector port
test voice port slot/subunit/port Injects a test tone into a voice port
inject-tone {local | network}
test voice port slot/subunit/port Performs loopback testing on a voice port
loopback {local | network | disable}
test voice port slot/subunit/port relay Tests relay-related functions on a voice port
test voice port slot/subunit/port Forces a voice port into fax mode
switch {fax | disable}
csim start xxxx Initiates simulated calls

A call must be established prior to all testing commands, except csim.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-149

 Verifying Voice Ports (Cont.)
Voice Port Testing Example

router#
test voice port inject-tone local 500hz
 Inject a 500-Hz test tone into the voice port
PSTN or
PSTN
PBX or
User
PBX User

1/0/1
PSTN or
PBX User

PSTN or
PBX User
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-150
 Summary

 Digital voice ports use TDM to carry multiple voice channels over a single
circuit.
 FXS ports connect gateways to end-user equipment, FXO ports to CO
switches or PBXs, and E&M ports to PBXs.
 Analog port settings define country-specific voice parameters, signaling
type, and in case of E&M, circuitry.
 Digital voice ports emulate analog signaling (CAS) or use CCS.
 ISDN uses CCS on BRI and PRI interfaces.
 Digital voice ports are defined by creating DS0 groups on the T1/E1
controller and configuring their signaling type.
 ISDN PRI interfaces are created by defining a pri-group on the T1/E1
controller.
 Tunable voice port parameters include cross-connects, timing, and comfort
noise.
 Echo cancellation is enabled by default but its time coverage can be tuned.
 Voice port and call parameters can be viewed.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-151

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-152
 Introduction to Voice
Gateways

Understanding DSP Functionality,

Codecs, and Codec Complexity

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-153

 Outline

 Voice Codecs
 Evaluating Quality of Codecs
 Evaluating Overhead
 Digital Signal Processors
 Codec Complexity
 Configuring Digital Signal Processors
 Verifying Digital Signal Processors

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-154

 Voice Codecs

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-155

 Voice Codecs
Codec Bandwidth
Codec Bandwidth [kb/s]
G.711/G.722 64
G.726r32 32
G.726r24 24
G.726r16 16
G.728 16
iLBC* 15.2, 13.3
GSM Full Rate (GSM-FR) 13
G.729 8
G.723r63 6.3
G.723r53 5.3

*iLBC = Internet Low Bitrate Codec

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-156
 Voice Codecs (Cont.)
G.729/G.729A/iLBC Comparison
 G.729/G.729A
– Both ITU standards, 8-kb/s, same compression delay (10 to
20 ms)
– G.729A less complex and processor-intensive, slightly worse
quality than G.729
– The Annex B variant can be applied to either codec
 Adds VAD and CNG
 iLBC
– Similar complexity but better quality than G.729
– Supports two fixed-bit-rate frame lengths:
 13.3 kb/s with an encoding frame length of 30 ms
 15.2 kb/s with an encoding frame length of 20 ms
– Supported on Cisco Unified IP phones and Cisco gateway
dial peers
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-157
 Voice Codec Packet Rates and Payload
Sizes

Codec Packets per Second (P/S) Payload Size [Bytes]

G.711 33 240
G.711 50 160
G.726r32 33 120
G.726r32 50 80
G.726r16 25 80
G.726r16 50 40
G.729 (a,b,ab) 33 30
G.729 (a,b,ab) 50 20
iLBC 33 50
iLBC 50 38
G.723r63 17 48
G.723r63 33 24
G.723r53 17 40
G.723r53 33 20

Low-bandwidth codecs produce samples at lower rate.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-158
 Evaluating Quality of
Codecs

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-159

 Voice Quality Evaluation
Test Methods
Mean opinion score (MOS):
 Defined in ITU-T Recommendation P.800
 Results in subjective measures
 Scores from 1 (worst) to 5 (best); 4.0 is business quality

Perceptual Evaluation of Speech Quality (PESQ):

 Automated assessment of the speech quality as experienced by users
 Successor of Perceptual Speech Quality Measurement (PSQM)
 ITU-T recommendation P.862 (Feb 2001)
 Worldwide applied industry standard for objective voice quality testing
 PESQ results principally model mean opinion scores (MOSs)

Perceptual Evaluation of Audio Quality (PEAQ):

 Automated assessment of speech and other audio types
 Patented and available under license
 PEAQ results principally model MOSs
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-160
 Voice Quality Evaluation
Test Methods Comparison

Feature MOS PSQM PESQ PEAQ

Test method Subjective Objective Objective Objective

End-to-end Inconsistent No Yes Yes

packet loss test

End-to-end jitter Inconsistent No Yes Yes

test

Measurement Voice and Voice Voice Voice and

subject other audio other audio

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-161

 Codec Quality

Codec Bandwidth [kb/s] MOS *

G.711 64 4.3
G.726r32 32 3.8
G.726r24 24 3.75
G.726r16 16 3.7
G.728 16 3.75
iLBC 15.2 4.14
GSM Full Rate 13 3.5
G.729 8 3.92
G.729a 8 3.7
G.723r63 6.3 3.7
G.723r53 5.3 3.65

*MOS values under ideal network conditions: no packet loss, low delay, and no jitter
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-162
 Evaluating Overhead

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-163

 Evaluating Overhead
Bandwidth Calculation
 The table lists Layer 3+ bandwidth per call, excluding overhead.
 More accurate bandwidth per call calculation would include
Layer 2 overhead.
Layer 3+
Packetization Voice Packets per
Codec Bandwidth
Period Payload Second
per Call
G.711 20 ms 160 Byte 50 80 kb/s
G.711 30 ms 240 Byte 33 74 kb/s
G.729 20 ms 20 Byte 50 24 kb/s
G.729 30 ms 30 Byte 33 19 kb/s

Bandwidth per call =

(Voice payload + Layer 3+ overhead + Layer 2 overhead) * packets per second * 8 bits/byte

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-164

 Evaluating Overhead (Cont.)
Layer 2 and Layer 3+ Overhead

Layer 2 Headers [Bytes]

802.3 Ethernet 18
802.1Q Ethernet 18+4
PPP 6-9
Multilink PPP with Interleaving 13
Frame Relay 6
Frame Relay with FRF.12 8

Layer 3 + Headers [Bytes] VPN Headers [Bytes]

IP 20 ESP 50-57
UDP 8 GRE/L2TP 24
RTP 12 MPLS label 4

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-165

 Evaluating Overhead (Cont.)
Bandwidth Calculation Example
 Example: Layer 3+, G.711 over Frame Relay, 50 Packets per
Second

Bandwidth per call

= (Voice payload + Layer 3 OH + Layer 2 OH) * packets per second x 8 bits/byte
= (160 + 40 + 6) bytes * 50 pps * 8 bit/byte
= 82,400 b/s = 82.4 kb/s

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-166

 Per-Call Bandwidth Using Common
Codecs

Call over
Voice Only Layer Call over
Codec PPS 802.3
Payload 3+ Frame Relay
Ethernet
G.711 160 bytes 50 80 kb/s 82.4 kb/s 87.2 kb/s
G.711 240 bytes 33 74.66 kb/s 76.27 kb/s 79.47 kb/s
G.729 20 bytes 50 24 kb/s 26.4 kb/s 31.2 kb/s
G.729 30 bytes 33 18.66 kb/s 20.27 kb/s 23.47 kb/s

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-167

 Digital Signal Processors

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-168

 Digital Signal Processors
Overview

DSPs
Analog or
Digital IP Packets
PSTN IP

Analog or
Digital IP Packets
PSTN

Speech IP Packets
IP

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-169

 Digital Signal Processors (Cont.)
Functions
 Voice termination
– Conversion between circuit-based voice and VoIP
– Echo cancellation, VAD, jitter management
 Media Termination Point (MTP)
– Passing one VoIP stream to another (same codec)
– Transformation between a-law and mu-law, or different
packetization periods
 Transcoding
– Conversion from one codec type to another
 Conferencing
– Mixing multiple voice streams

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-170

 Digital Signal Processors (Cont.)
Branch

Media
Termination
Central site
Transcoding
and/or IP WAN
Conferencing Conferencing
G.729
IVR
G.711

PSTN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-171

 Digital Signal Processors (Cont.)
DSP Chip
 The DSP chip performs the sampling, quantization, encoding,
and optional compression.
 The number of simultaneous calls that a chip can handle
depends on the type of DSP and the codec being used.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-172

 DSP Modules
PVDM2 and PVDM3 Overview
PVDM2 installed in:
 Motherboard PVDM2 slot on Cisco 2800 and 3800 Series
ISRs
 Cisco high-density digital voice network modules (NM-
HDV2, NM-HDV2-1T1/E1 and NM-HDV2-2T1/E1)
 PVDM2 Adapter for PVDM3 slot on Cisco 2900, 3900
Series ISRs

PVDM3 installed in:

 Motherboard PVDM3 slot on Cisco 2900, 3900 Series ISR
routers
– Cisco 2901 and 2911 routers have 2 slots each, Cisco
2921 and 2951 routers have 3 slots each, and Cisco
3925 and 3945 routers have 4 slots each.
– Cisco IOS Software Release 15.0.1(M) and later

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-173

 DSP Module Comparison
PVDM2 / PVDM3 Comparison

PVDM2 PVDM3
Platform Cisco 2800, 3800, 2900, Cisco 2900,3900 Series ISRs
support 3900 Series ISRs
Models PVDM2-8, PVDM2-16, PVDM3-16, PVDM3-32, PVDM3-64,
PVDM2-32, PVDM2-48, PVDM3-128, PVDM3-192, PVDM3-256*
PVDM2-64*
Capabilities Voice/Fax Voice/Video (No Cisco Fax Relay)
Resource Per-module and per-chassis DSP resources in motherboard slots
sharing sharing shared across the chassis backplane
Coexistence Can coexist on the Cisco 2900 and 3900 Series ISR platforms but
PVDM2 cannot be installed directly on the motherboard

* Number in the model name identifies the number of supported G.711 channels
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-174
 Codec Complexity

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-175

 Codec Complexity
Media Termination and Transcoding
Low complexity Medium complexity High complexity
G.711 and Clear- G.729A, G.729AB, G.723.1, G.728, G.729,
Channel Codec G.726, G.722, and G.729B, iLBC, Modem
Fax Relay Relay
PVDM2-8 8 4 4
PVDM2-16 16 8 6
PVDM2-32 32 16 12
PVDM2-48 48 24 18
PVDM2-64 64 32 24
PVDM3-16 16 12 10
PVDM3-32 32 21 14
PVDM3-64 64 42 28
PVDM3-128 128 96 60
PVDM3-192 192 138 88
PVDM3-256 256 192 120

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-176

 Codec Complexity (Cont.)
Recommended Usage in Deployment Models

Deployment model Recommended Description

codec
Single-site deployment • G.711/G.722 • Codec used for VoIP calls within the
same site where enough bandwidth is
available.
Multisite WAN with • G.711/G.722 for • Intrasite calls consume more bandwidth
centralized call intrasite calls and provide the best voice quality.
processing, distributed • G.729/A/B for • Intersite calls consume little bandwidth
call processing, and intersite calls and provide good voice quality.
clustering over the IP
WAN

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-177

 Packet Voice DSP Module Conferencing
Conferencing
 PVDM3 allows sharing the same DSP between transcoding, voice
termination, and conferencing.
 PVDM2 allows sharing the same DSP for voice termination and
transcoding, but a dedicated DSP is needed for conferencing.
 The number of supported conferences and participants depends on
codec complexity. For example, PVDM3-256 supports:
– 66 G.711 conferences with 8 participants each
– 6 G.711 conferences with 64 participants each
– 30 G.722 conferences with 8 participants each
– 36 G.729/G.729A conferences with 8 participants each
– 18 iLBC conferences with 8 participants each
– Up to 32 participants per G.729/G.729A/G.722 conference
– Up to 16 participants per iLBC conference

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-178

 DSP Calculator

Select the Router

Model

2
Select the Cisco IOS
Software Release

Cisco.com: http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-179
 Configuring DSPs

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-180

 Configuring DSPs for Voice Termination
Configurable Codec Complexity Options
 High
– Support a high complexity codec or combination of high and lower
complexity codecs
 Medium
– Support a medium complexity codec or combination of medium and low
complexity codecs
– Greatest number of voice channels
 Flex
– Allows oversubscription—more voice channels can be configured or
connected to the module than the DSPs can accommodate
– If all voice channels go active simultaneously, some are unable to
allocate a DSP resource
– Default setting
 Secure
– Supports SRTP package capability
– Lowest number of selected medium-complexity calls per DSP

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-181

 Configuring DSPs for Voice Termination
(Cont.)
Configuring Voice Card
router(config)#
voice-card slot
 Enters the voice card configuration mode

router(config-voicecard)#
dspfarm
 Adds voice card to a DSP resource pool
 Local DSPs are available for TDM streams on a different module or VWIC

router(config-voicecard)#
codec complexity {flex | high | medium | secure}
 Specifies codec complexity

router(config-voicecard)#
codec sub-sample
 Doubles the sampling frequency for G.711

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-182

 Codec Complexity Configuration
Voice Card Configuration Example

router(config)# voice-card 1
router(config-voicecard)# codec complexity ?
flex Set codec Flex complexity, higher call density.
high Set codec to high complexity, lower call density.
medium Set codec to mid range complexity and call density
secure Set codec complexity to secure.
router(config-voicecard)# codec complexity flex
router(config-voicecard)# codec sub-sample

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-183

 Configuring DSP Resources for
Transcoding, Conferencing, and MTP
DSP-Farm Configuration
router(config)#
voice-card slot
 Enters the voice card configuration mode

router(config-voicecard)#
dsp services dspfarm
 Enables DSP farm services
 Makes the DSP services available for conferencing and transcoding

router(config)#
dspfarm profile profile-identifier {conference|mtp|
transcode}
 Creates a DSP farm profile for conferencing, MTP, or transcoding

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-184

 Configuring DSP Resources for
Transcoding, Conferencing, and MTP
DSP Farm Profile Configuration
router(config-dspfarm-profile)#
codec codec-type
 Specifies the allowed codecs

router(config-dspfarm-profile)#
maximum sessions number
 Defines the maximum number of sessions

router(config-dspfarm-profile)#
associate application sccp
 Enables SCCP for the profile

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-185

 Transcoding and Conferencing
Example
Configuration Example

Cisco Unified
Communications
Manager IP WAN
10.1.1.201

PSTN

dspfarm profile 1 transcode dspfarm profile 1 conference

codec g711ulaw codec g711ulaw
codec g711alaw codec g711alaw
codec g729ar8 codec g729ar8
codec g729abr8 codec g729abr8
codec g729r8 codec g729r8
maximum sessions 6 codec g729br8
associate application SCCP maximum sessions 2
no shutdown associate application SCCP
no shutdown
© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-186
 Verifying DSPs

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-187

 Verifying DSPs
router# show voice dsp
DSP DSP DSPWARE CURR BOOT PAK
TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK
COUNT
==== === == ======== ========== ===== ======= === == ========= ================
edsp 0001 01 g711ulaw 0.1 busy 50/0/1.1
edsp 0002 02 g729r8 p 0.1 IDLE 50/0/1.2
edsp 0003 01 g711ulaw 0.1 IDLE 50/0/2.1
edsp 0004 02 g729r8 p 0.1 IDLE 50/0/2.2

----------------------------FLEX VOICE CARD 0 ---------------------------

*DSP VOICE CHANNELS*
CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending
LEGEND : (bad)bad (shut)shutdown (dpend)download pending

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK
COUNT
====== === == ========= ===== ===== ======= === == ========= == ==== =========
C5510 001 01 g711ulaw 26.6.0 busy idle 0 0 0/0/0:15 08 0 228/9228
*DSP SIGNALING CHANNELS*
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK
COUNT
====== ===Inc. All==
© 2010 Cisco Systems, rights =========
reserved. ========== ===== ======= === == ========= == ==========
CVOICE v8.0—1-188
 Verifying DSPs (Cont.)
Verifying DSP Farm

Router# show dspfarm dsp all

SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0 5 1.0.6 UP N/A FREE conf 1 - - -

0 5 1.0.6 UP N/A FREE conf 1 - - -

Total number of DSPFARM DSP channel(s) 2

Two Conference
Bridges Configured

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-189

 Summary

 Codecs compress and decompress data when converting

voice signals to VoIP packets.
 The MOS scores of major codecs represent their perceived
quality and range between 3.5 and 4.3.
 Total bandwidth of one VoIP conversation depends on the
packet ratio, codec, and Layer 2-to-Layer 4 overhead.
 DSPs are specialized voice processors used for voice
termination, conferencing, transcoding, and MTP.
 The number of voice channels served by one PVDM
depends on the codec complexity: low, medium, or high.
 Media resources available for voice termination can be fine-
tuned for codec complexity.
 The status of media resources can be verified on the voice
gateway.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-190

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-191
 Module Summary

 Voice gateways support the Cisco Unified Communications

architecture by converting voice signals and offering
advanced voice features.
 Call routing involves incoming and outgoing call legs that
correspond to inbound and outbound dial peers.
 Gateways support various interface types: analog with in-
band signaling (FXO, FXS, FXS-DID, E&M), digital with CAS
signaling (T1/E1 CAS), and digital with CCS signaling (T1/E1
PRI, BRI).
 Voice conversion into VoIP uses codecs with varying
complexity and MOS, and is performed by dedicated DSPs.

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-192

© 2010 Cisco Systems, Inc. All rights reserved. CVOICE v8.0—1-193