Introducing Voice Gateways Part 1

Introducing Voice
Gateways Part 1
Chapter 1
Pages 1 to 57
Introducing Voice Gateways Part 1
Josh Lowe Winter 2013
THE ROLE OF GATEWAYS
Traditional Telephony Networks

Telephones Analog phones connect directly to the PSTN
Central Office (CO) Switch Terminate local loops, provide
signaling, call setup, etc..
Private Branch Exchange Smaller, privately owned version
of a CO Switch
Trunk Used to connect two switches together
CO Trunk Connects local CO switch to a PBX
Tie Trunk Directly connects to PBXs
Interoffice Trunk Connects two COs

Traditional telephony systems address:
Signaling: generate and exchange control information between
two endpoints
Database Services: provides access to additional information
and services
Bearer Control: defines and controls the bearer channels that
will carry the actual voice calls
Cisco Unified Communications

What do we mean by Unified Communications?
Traditionally data, voice, and sometimes video each
had separate networks, including separate staff to
manage them
Different communications processes were handled with
different systems (e.g. a mail server for e-mail, a
voicemail server for VM, an IM server for instant
messaging, etc.)

Unified Communications is an amalgamation of
various networks and communication services
Uses a single network to carry data, voice and video

This single network approach allows us to unify our
communications
For example you can now receive your VM in your email inbox, or listen to your e-mails over the phone

Architecture
Cisco Unified Communications Manager

Cisco Unified Communications Manager (CUCM)
provides the brains of Cisco VoIP networks
Provides device control, call routing, permissions,

features and connectivity
Comes in three variations (currently)
Cisco Unified Communications Manager Express (CUCME)
Cisco Unified Communications Manager Business Edition
(CUCMBE)
Previously called CallManager. Many people (myself

included) still refer to it as CallManager on a regular
basis

Full-featured, allows for clusters with a limit of up to 30,000
phones per cluster (but you can have multiple clusters)
Server-based
Web-based GUI
Configuration
CUCM Cluster
10

Cisco Unified Communications Manager Express
(CUCME)
Small-office, branch office design
Runs on Cisco ISR and ISRG2 routers
Supports 35 to 450 phones depending on the router
IOS-based configuration (but can be configured with various
GUI front-ends)
Includes many of the features available in CUCM

Can also run in Survivable Remote-Site Telephony (SRST)
mode where it acts as a failover system for branches connected
to a CUCM cluster across a WAN
11

Cisco Unified Communications Manager Business
Edition (CUCMBE)
Designed for medium businesses with up to 350 users and up
to 20 sites
Single-server solution for Voice, Video, Messaging, Mobility and
Presence
Server-based
Web-based GUI configuration
12

Gateways
What is a gateway?
A gateway is a device that connects two different voice
networks together
Can be any or all of:

A device which acts as a voice switch that interconnects
multiple traditional telephony circuits
A device which acts as a PSTN-to-VoIP gateway that provides
translation between VoIP and non-VoIP networks, such as the
PSTN
A device which acts as a Cisco Unified Border Element (CUBE)

that interconnects two IP networks and allows communications
between endpoints distributed among them
17
Gateway Operation
Gateways operate using several control and callsignaling protocols
H.323
Media Gateway Control Protocol (MGCP)
Session Initiation Protocol (SIP)
Skinny Client Control Protocol (SCCP, sometimes called
Skinny)
18
H.323
H.323 is actually a suite of protocols defined by the
ITU-T. It is a peer-to-peer protocol
Most widely deployed standards-based voice and video

conferencing standard for IP networks
Includes the following protocols:
H.225 Call Signaling: Establishes connections between two
H.323 endpoints or an endpoint and a gatekeeper
H.225 Registration, Admission, and Status (RAS): Performs
registration of endpoints, call admission control, bandwidth
changes, status monitoring, etc.
H.245 Control Signaling: End-to-end control signals
Audio and Video Codecs: More on these later

19
MGCP
MGCP is a centralized client-server call control protocol
Centralized gateway administration provides largely
scalable VoIP networks
All dial plan information resides on a separate call

agent which controls the ports on the gateway (router)
and performs call control
In Cisco-based network, Cisco Unified Communications
Manager acts as the call agent, controlling the gateway
(router)
Note that although a Cisco UCM Express router can act
as an MGCP gateway, it cannot be a call agent. You
must also have a full Cisco UCM cluster deployed
20
MGCP
With MGCP, Cisco UCM knows of, and controls,
individual voice ports on an MGCP gateway
Uses plain-text commands sent over UDP port 2427

between the Cisco UCM (call agent) and a gateway
Because the call agent has all the call routing
intelligence, you do not need to configure the gateway
with all the dial peers it would otherwise need
The downside is that a call agent must always be

available to service call requests (although SRST and
H.323 can be used to provide emergency routing in
case of a WAN outage)
21
SIP
SIP is an open standard based on the logic of the
World Wide Web and very simple to implement
Widely used with gateways and proxy servers within

service provider networks
SIP is a peer-to-peer protocol where user agents (UA)
initiate sessions, similar to H.323
Unlike H.323, SIP uses ASCII-text-based messages to
communicate, very easy to implement and troubleshoot
Cisco UCM does not control SIP devices because of

the peer-to-peer nature of SIP
22
SCCP
SCCP (skinny) is a Cisco proprietary protocol used to
communicate between Cisco UCM and terminal
endpoints (such as IP phones)
Client/server protocol meaning any event (such as onhook, off-hook, or button presses) sends a message to
a Cisco UCM
Cisco UCM then tells the device what to do about the

event
SCCP is the most common protocol used with Cisco IP
phones
SCCP can interoperate with H.323-compliant terminals
SCCP is not normally used for inter-site VoIP (LAN

only)
23
Comparing VoIP Signaling Protocols

The primary goal for all four of the previously
mentioned VoIP signaling protocols is the same: to
create a bidirectional RTP stream between VoIP
endpoints
Each protocol uses different architectures and

procedures to achieve this goal
24
H.323
Gateway configuration is somewhat complex because
you need to define the dial plan and route patterns
directly on the gateways
H.323 is responsible for all the signaling between a
Cisco UCM cluster and H.323 gateway.
ISDN protocols, Q.921 and Q.931, are used only on

ISDN links to the PSTN
25
MGCP
Gateway configuration is simpler because the dial plan
and route patterns are defined on the Cisco UCM
cluster instead of directly on the gateway
All ISDN Layer 3 information is backhauled to Cisco
UCM
Only ISDN Layer 2 information (Q.921) is terminated on

the gateway
26
SIP
Like H.323, SIP is a peer-to-peer protocol, making
gateway configuration relatively complex because the
dial plan and route patterns need to be defined on the
gateways
SIP is responsible for the signaling between Cisco

UCM and the gateway
Q.921 and Q.931 are used only on ISDN links to the
PSTN
27
SCCP
SCCP is a client/server protocol which simplifies the
configuration of SCCP devices, similar to MGCP
Standard analog and digital signaling can be used to

reach an analog device directly connected to the
gateway or to reach the PSTN
28
Gateway Deployment Example
29
IP Telephony Deployment Models

Single-site
Multisite with centralized call processing
Multisite with distributed call processing
Clustering over the IP WAN

Each model differs in the type of traffic that is carried
over the WAN, the location of the call-processing agent,
and the size of the deployment
30
Single-Site Deployment
Call processing done at a single site or campus

No telephony services provided across the WAN (data
only)
31
Design Characteristics of Single-Site

Deployment
Single Cisco UCM cluster
PSTN for all calls outside the site
Easy to deploy
Simplified dial plan

Bandwidth typically not an issue
No transcoding resources are required because of the
use of a single high-bandwidth codec
All Cisco UCM servers, applications, and DSP

resources are located in the same physical location
32
Multisite WAN with Centralized CallProcessing Deployment

Single call-processing
cluster provides services to
multiple remote sites using
the IP WAN
33
Design Characteristics of Multisite WAN with

Centralized Call-Processing Deployment
Single UCM cluster hosted at central location
Remote sites rely on centralized CUCM cluster to handle
their call processing
WAN used for inter-office calls when possible
WAN used to transfer CUCM traffic between sites (e.g.

phone registration messages)
PSTN for all external calls
QoS and call admission control required on WAN to ensure
call quality between offices
Applications (such as voicemail) are usually hosted at the

central site as well
SRST can be used on the remote sites in case the WAN
goes down
34
Multisite WAN with Distributed CallProcessing Deployment
Multiple independent sites each with its own callprocessing cluster connected to an IP WAN
35
Design Characteristics of Multisite WAN with

Distributed Call-Processing Deployment
IP WAN connects all the distributed call-processing
sites, with the PSTN typically serving as a backup
connection
Each site is completely self-contained and in the case
of a WAN failure a site does not lose call-processing
service or functionality, calls are simply sent between
the sites over the PSTN
Cost savings when you use the IP WAN for calls
between sites
Maximum utilization of available bandwidth by allowing
voice traffic to share an IP WAN with other types of
traffic
Scalable to hundreds of sites
36
Clustering over the IP WAN Deployment
Applications and Cisco UCM of the same cluster are

distributed across the IP WAN
37
Clustering over the IP WAN Deployment

Clustering over the WAN can support two types of
deployments:
Local failover deployment model:
UCM subscriber and backup servers located at the same site
No WAN between subscriber and backup servers
Ideal for two to four sites with UCM.
Remote failover deployment model:
Deploy the backup servers over the WAN

Up to eight sites with UCM subscribers backed up by UCM
subscribers at another site.
A combination of the two models can also be used to

satisfy specific requirements
38
Benefits of the Clustering over the IP

WAN Deployment
Single point of administration for users for all sites
within a cluster
Feature transparency
Shared line appearances across sites
Extension mobility across sites
Unified dial plan
39
WAN Considerations for Clustering over

the IP WAN Deployment
Delay
The maximum one-way delay between any UCM servers should
not exceed 40ms (80ms round-trip)
Jitter
Variable delay, should be minimized using QoS
Packet loss and errors

QoS should be used to prioritize VoIP traffic, but Intra-Cluster
Communication Signaling (ICCS) uses TCP so should not be a
huge factor
Bandwidth
Provision bandwidth based on expected call volume and the
types and number of devices. Take into consideration other
applications using the WAN
40
Modern Gateway Hardware Platforms

Cisco 2900 Series Integrated Services Routers G2
Cisco 3900 Series Integrated Services Routers G2
41
Well-Known Older Enterprise Models

Cisco 2800/3800 Series Integrated Services Routers
Some Cisco 2600XM modular routers can also be used

for VoIP
42
Specialized Voice Gateways

Cisco Analog Telephone Adaptor (ATA) 186
Used to connect traditional telephony devices (especially fax

machines) to the VoIP network.
Has two analog voice ports and an Ethernet port on the back
Cost effective!
43

Cisco VG248 Analog Phone Gateway
Provides 48 analog ports to connect traditional devices to the
VoIP network
44

Cisco AS5300XM/AS5400XM Series Universal
Gateway
All-in-one gateway that provides data, voice, and faxes
services.
Supports PSTN signaling, gateway signaling (H.323, SIP, etc),
voice codecs, Call Admission Control, etc.
Session Border Controller (SBC) allows different gateway
protocols to communicate (SIP-to-H.323, H.323-to-SIP, etc)
45

Cisco Unified Communications 500 Series
All-in-one unified communications device, UC520 Includes:
Cisco UCM Express (CUCME) for IP phone services

Cisco Unity Express (CUE) for voicemail services
Integrated 8-port PoE LAN switch
Network security features (Cisco IOS Firewall, VPN, etc)

Optional integrated Wireless LAN Controller module
And more!
46
Gateway Operational Modes

Voice-switching gateway
Connects analog-to-analog, analog-to-digital, and digital-todigital voice circuits (no IP, just traditional voice)
Equivalent to a PBX or central office (CO) switch
47

VoIP Gateway
Connects traditional telephony networks and equipment to the
IP network.
Converts signaling and media transmission methods from one
side to the other
Provides physical access for local analog/digital devices like
telephones, fax machines, key sets, and PBXs
48

Cisco Unified Border Element (Cisco UBE, or
CUBE)
Sometimes referred to as an IP-to-IP gateway or a session
border controller (SBC), it connects two IP networks together.
For example, a CUBE would connect an H.323 network to a SIP
network, or a SIP network to a different SIP network
Terminates the signaling for each network and either passes
the media through or terminates and reinitializes the media
channel
49
Call Legs
Voice calls are segmented into discrete sections called
call legs. Call legs are router-centric
The incoming part of the call is referred to as the

incoming call leg
The outgoing part of the call is referred to as the
outgoing call leg
50
Call Legs
On Cisco IOS gateways the call legs are associated
with dial peers (much more on this later!)
One dial peer corresponds to one call leg

If the gateway receives or forwards a call over an
analog or digital voice circuit the call legs is referred to
as POTS (Plain Old Telephony Service) and the dialpeer is a POTS dial peer
If the gateway receives or forwards a call over an IP
network the call legs is referred to as VoIP and the dialpeer is a VoIP dial peer
51
Call Legs
Call legs are essential to call routing
Gateways apply settings defined in the incoming call
leg before making call-routing decisions
These settings can determine how dialed digits are

collected and how call routing patterns are matched
For example, an incoming call can have the destination
telephone number changed on the incoming call leg so
that the gateway forwards the call to a different location
For VoIP call legs, these settings can include things like
voice codecs to use and which VoIP features to enable.
Often these parameters must match between the local
and remote gateways before a call is forwarded
52
Call Legs: Voice-Switching Gateway

Uses traditional telephony interfaces
Receives and forwards the call setup request over
analog or digital voice circuits
May have to convert the call signaling and voice format

when the call moves from one port to another.
Uses POTS for both the incoming and outgoing call
legs
53
Call Legs: VoIP Gateway

Provides translation between VoIP and non-VoIP
networks such as the PSTN
Converts the signaling between traditional telephony

circuits and VoIP transmissions of an IP network
One call leg is a POTS call leg and the other is a VoIP
Call leg
54
Call Legs: Cisco Unified Border Element

Forwards incoming VoIP calls to another VoIP network
Incoming VoIP signaling method may be different than
outgoing VoIP signaling (e.g. H.323 to SIP)
Actual voice media channel may be either flowthrough or flow-around

Both call legs are VoIP call legs
55
HOW VOICE GATEWAYS

ROUTE CALLS
56
Gateway Call-Routing Components

Call routing is based on dial peers
Dial peers identify the source and destination endpoints
of a call
Define the characteristics applied to each call leg in the

connection
57
Gateway Call-Routing Components

When an inbound call arrives on a gateway, the
gateway finds the inbound dial peer and processes its
settings.
If the settings are acceptable, the gateway finds the
outbound dial peer, establishes the outgoing call leg,
and the call is switched from the incoming call leg to the
outgoing call leg
You need to configure dial peers to enable call routing

on a gateway
58
Dial Peers
Example: This gateway has two dial peers, a POTS dial
peer for the analog phone and a VoIP dial peer for the
VoIP network.
59
Dial Peers
The POTS dial peer configuration includes (at a
minimum) the telephone number of the analog phone
and the voice port to which it is attached
60
Dial Peers
The VoIP dial peer is connected to the IP network and
includes (at a minimum) the destination phone number
(or a range of numbers) and the next-hop IP address
61
Dial Peers
62
IP and Call Routing Comparison
63
Secondary Paths
Often call routing involves multiple paths, such a an IP
WAN with a PSTN backup
64
Configuring POTS Dial Peers

Consider the following example network:
65

Router 1 configuration:
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
66

Router 2 configuration:
R2(config)# dial-peer voice 1 pots
No forwarddigits
necessary!
R2(config-dialpeer)# destination pattern 2001

R2(config-dialpeer)# port 1/0/0
67
Matching a Dial Peer

Two telephone numbers are typically sent with the call:
The calling number, also known as the Automatic Number
Identification (ANI). This is the number of the person who is
placing the call; the source.
The called number, also known as the Dialed Number

Identification Service (DNIS). This is the number of the
person who is being called; the destination
Both of these numbers can be used to find the inbound

and outbound dial peers
68

Dial peers are matched based on one of:
The incoming voice port (e.g. port 1/0/1)
The destination-pattern command (e.g. destinationpattern 2001)

The incoming called-number command (e.g. incoming
called-number 2001)
The answer-address command (e.g. answer-address 1001)
69

destination-pattern
command
Matches the outbound dial peer based on the called number
Also can be considered when matching the inbound dial peer,

but is matched based on the calling number instead
incoming called-number
command
Only considered when matching the inbound dial peer and

matches on the original called number
answer-address
command
Only considered when matching the inbound dial peer and

matches on the original calling number
70
Number Matching
Destination-patterns, incoming called-number, and
answer-address can all use wildcards to define the
numbers they match.
They dont have to explicitly match a destination
number
This is useful if you have a range of phone numbers or

extensions that are all routed to the same gateway or
endpoint
For example, extensions 3001 to 3050 all go out voice
port 1/0/0
71
Regular Expressions Used to Match

Number Strings
72
Number Matching Examples

Number
Matching Telephone Number(s)
5551234
Matches one telephone number exactly, 5551234.

Typically used when there is a single device, such as a
telephone or fax machine, connected to a voice port.
^555
Matches any number starting with 555 using an explicit

match of the beginning of the string.
555123[5-9]
Matches the number range 5551235 through 5551239.
55512[3-4].
Matches a seven-digit telephone number where the first

five digits are 55512, the sixth digit can be a 3 or 4, and the
last digit can be any digit. (Notice the dot at the end?)
Used when telephone number ranges are assigned to

specific sites.
73
Number Matching Examples

Number
Matching Telephone Number(s)
.T
Matches any telephone number that has at least one digit

and can vary in length from 1 through 32 digits.
Used for a dial peer that services a variable length
dial plan for local, national, and international calls.
Also used as a default destination pattern so that any calls
that do not match a more specific pattern will match this
pattern and can be directed to an operator.
(200)?5551234 Matches the numbers 2005551234 and 5551234. This

expression uses a pattern (200) that can occur zero or one
time.
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.
74
Inbound Dial Peer Matching Order

The gateway selects an inbound dial peer by matching
the information elements in the setup message with the
dial-peer attributes in the following order:
1. Called number with incoming called-number.
2. Calling number with answer-address.

3. Calling number with destination-pattern.
4. Voice port (the port where the incoming call is received)
5. Default dial peer
The gateway stops searching when a dial peer match is

found (this is important to know for troubleshooting)
75
Inbound Dial Peer Matching Example
Match (outbound)
Outbound dial peer on R1
(inbound dial peer not shown)
Match (inbound)
No Match
Match (inbound)
76
Inbound Dial Peer Matching

Remember that the answer-address command matches
on the calling number (the source)
Use the answer-address command when matching the

geographical region of the caller
For example, callers from a given country could be
redirected to the appropriate language-speaking team
Or, callers from a specific region could be directed to
the regional sales staff
77
Inbound Dial Peer Matching

Use the incoming called-number command whenever
possible
All types of call setup messages and signals always

include the DNIS information (called number), but dont
always carry the ANI information (calling number)
78
Matching Outbound Dial Peers

Outbound dial peers are much simpler than inbound
dial peers because they only match on the
destination-pattern command
The incoming called number (DNIS) is used to match a
destination-pattern on an outbound dial peer
Both POTS and VoIP dial peers are considered equally

for outbound dial peer matching.
An outbound POTS dial peer uses the port command
to forward the call out an analog or digital port
An outbound VoIP dial peer uses the session target
command to forward the call to a VoIP peer
79
Matching Outbound Dial Peers
80
Default Dial Peer

The figure depicts a situation in which the call routing
works only in one direction
What inbound dial peers would be selected on each

gateway?
Matches
Outbound
Matches
Outbound
81
Default Dial Peer

If no inbound peer can be matched by the defined
criteria, the gateway resorts to the default dial peer
The default dial peer is referred to as dial peer 0 and is

used for inbound matches only. It will never match
outbound call legs.
The characteristics of dial peer 0 cannot be changed
82
Default Dial Peer

The default characteristics of dial peer 0 for VoIP peers
are:
G.729 and G.711 codecs are supported
IP precedence is set to 0 (for QoS)
VAD is enabled
RSVP is not supported (for QoS)
Fax-rate service is supported
The default characteristics of dial peer 0 for POTS

peers are:
No applications are supported.
No direct inward dialing (DID) is supported.

83
Default Dial Peer

Avoid using the default dial peer whenever possible
Many problems with calling out through a Cisco IOS
gateway are caused by codec, VAD, and DTMF-relay
misconfigurations when dial peer 0 is being matched
Having the incoming called-number command
configured correctly ensures that the dial peer is always
matched with the parameters that you want when
placing calls through a gateway
Remember you can also match inbound using the
answer-address command, the destination-pattern
command, and the incoming port (for POTS)
84
Direct Inward Dialing

In the early days of traditional telephony, enterprises
used two-stage dialing to allow outside callers to
reach internal telephones
When an inbound call was received the CO switch
seized the voice port and the PBX presented another
dial tone (called a secondary dial tone) to the caller
The caller would then dial the extension of the internal
telephone
These days regular two-stage dialing would be very
confusing (and annoying) to callers who are used to
one-stage dialing
85

One-stage dialing was made possible by the invention
of direct inward dialing (DID)
With one-stage dialing, the caller enters the entire

called-party number, including the number required to
reach the internal telephone
The PSTN CO switch sends either the entire DNIS or
just a section of the number to the PBX, which forwards
the call to the internal telephone
The caller does not hear a secondary dial tone and
does not have to dial any further digits
86

DID must be enabled on inbound POTS dial peers
It is supported on all digital voice ports (T1/E1/ISDN)
and on analog FXS-DID ports
It is not supported on any other analog ports (FXS,

FXO, E&M)
87
Two-Stage Dialing Example

Step 1: The user takes the phone off-hook, receives
the dial tone from the PSTN CO switch, and dials 5552001
555-2001
88

Step 2: The PSTN receives the digits and delivers the
call to the destination gateway. The gateway does not
receive the number that was dialed by the caller
555-2001
89

Step 3: The caller hears a secondary dial tone sent by
the gateway at the destination. The caller dials
extension 2001
555-2001
90

Step 4: The gateway collects the rest of the number
one digit at a time and uses the number 2001 to match
the outbound dial peer, signaling a call on port 1/1/1
and the telephone rings
555-2001
91
One-Stage Dialing
One-stage dialing is enabled when the DID feature is
configured on the inbound POTS dial peer of the
destination voice gateway
The caller enters the entire number without a
secondary dial tone
Note that telephony service providers often charge a

fee for DID trunks
92
One-Stage Dialing
Step 1: The user takes the phone off-hook, receives
the dial tone, and dials 555-2001
93
One-Stage Dialing
Step 2: The PSTN delivers the call to the destination
gateway, usually stripping off most of the digits. The
destination gateway receives the last four digits of the
called number (2001) in one call setup message or over
an analog FXS DID trunk
94
One-Stage Dialing
Step 3: The destination gateway matches the outbound
dial peer and signals an incoming call to port 1/1/1. The
recipient phone rings.
95
Configuring DID
The string . (single period) matches any number with at
least one digit. This allows dial peer 1 to be the inbound
dial peer for any number dialled
The direct-inward-dial command enables DID for
this dial peer, allowing one-stage dialing
96
Configuring DID
The string .T (single period followed by T) matches any
number with at least one digit. Its used here as the
outbound dial-peer, sending all received calls out port
1/0/0.
The .T string is used here as an example, but in reality

should only be used in very particular (and rare)
circumstances
97
98

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