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[4]
INTRODUCTION CCVP GWGK Quick Reference Sheets by Brion S. Washington
Introduction
For as long as anyone alive can remember, there has always been a way Telecommunication monopolies have been declining for years.
to pick up some sort of phone for chats with distant relatives or with Everywhere you look, smaller companies are evolving, while creating
companies that offer customer support. The protocols for this have been new services and functionality for voice communications. A safe guess
ever changing since the invention of the telephone. The old days of is that a few hundred billion phone calls are completed every year in
human operators gave way to central office switches, which in turn saw the world. With the customer pool consisting of hundreds of millions,
the uptake of private branch exchange (PBX) use in companies. companies are competing to add services and growing their client base.
Telecommunication methods are ever changing to meet customer Because smaller companies can now enter the market and challenge the
expectations and to generate new services (and thus money). established companies, proprietary networks of the past will be just
that: of the past.
In the early stages of voice communications, the rise of major telco
companies brought numerous changes. For the most part, the changes History shows that customer demand drives technology and new inven-
proved to be beneficial to the growth of the industry. Among these tions. One demand has been for the intertwining of voice and data
benefits was the financial power to expand services to all parts of the networks. But, most companies must answer to shareholders, and
world. Then, no matter where you lived, you probably had access to a shareholders want a return on investment (ROI). The demand for the
phone. This almost universal access allowed providers to generate more intertwining of voice and data networks, combined with the need to cut
revenue and reinvest in emerging technologies and research. The down- costs (and thus increase ROI), has helped spark the Voice over IP
side to this was that small companies could not compete with the giants (VoIP) revolution.
of the field. When the smaller companies couldn’t get a foothold into
Modern telecommunications use a variety of protocols and devices.
the market, larger companies continued their dominance. This market
lockout started a selfish trend: Companies could do what they wanted VoIP was one of the technologies created to help combine voice and
with their equipment with little regard for client needs or interoperabil- data networks. VoIP is a method of sending voice and video over data
networks. VoIP has emerged as the de facto way for companies to send
ity with other companies. This monopoly created an environment in
which the equipment was proprietary. Proprietary equipment, such as a calls. Within VoIP, there are a few protocols to choose from, as follows:
PBX, from one company is probably not going to be interoperable with n H.323
another company’s PBX. The purchasing company is now tied to one
n SIP (Session Initiation Protocol)
brand of equipment.
© 2007 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 63 for more details.
CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets By Brion S. Washington ISBN: 1587053241 Prepared for Duc Le, Safari ID: anhdle@CISCO.COM
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[5]
PURPOSE OF THIS GUIDE CCVP GWGK Quick Reference Sheets by Brion S. Washington
n MGCP (Media Gateway Control Protocol) As a tester, you must be able to decide which functions to configure
n SGCP (Simple Gateway Control Protocol) based on the requirements. It is not this guide’s intention to try to
explain or even mention all the possible options that can be configured
n H.248/MEGACO with Cisco GKs and GWs. Instead, I try to give you the information
Each of these protocols has its advantages and disadvantages. The you must have, presented in such a way that you gain a full understand-
protocols have different components to provide services. The compo- ing of the subject matter.
nents can be grouped by the function they serve. Examples of these
components include gatekeepers (GK), gateways (GW), endpoints,
circuits, call control, and different signaling types. Those who want to Who Should Read This Guide?
implement some type of VoIP need to understand which protocol best Anyone who wants to augment his or her study material for the GWGK
suits their voice needs. exam is the target audience for this guide. However, my intention also
is to provide a deeper level of understanding for those who want to read
a concise document to gain a fundamental understanding of GW and
Purpose of This Guide GK operation in Cisco environments. This guide does not delve into
This guide is intended to help readers pass the Gateway Gatekeeper GK/GW minutiae. Instead, it provides a solid foundation on which you
(GWGK) exam required for the Cisco Certified Voice Professional can build. For those readers who plan to prepare for additional certifi-
(CCVP) and other voice-related certifications. The CCVP certification cations, this short cut could supplement your study material for other
is valid for three years. After passing the CCVP, you may continue on voice-related certifications (including CCIE Voice).
to Cisco Certified Internetwork Export (CCIE) Voice certification. I
hope readers find this guide useful as a preparation tool if they choose —Brion S. Washington
to pursue the CCIE Voice certification.
In this guide, you will find the necessary information to prepare for
your exam, as outlined at http://www.cisco.com. The 642-452
Implementing Cisco Voice Gateways and Gatekeepers exam tests your
ability to configure and set up various functions on these Cisco devices.

[6]
GATEKEEPERS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Gatekeepers When GKs are used in your H.323 network, GKs must/may provide
certain mandatory and optional functions. Some of the functions are
In the VoIP world of H.323, the GK plays a vital role if included in the automatically configured when the GK service is started; others must
network. The use of a GK is optional; if chosen, the GK must perform be configured manually.
certain functions. Gatekeepers act like the brains in an H.323 network.
They provide network access, bandwidth management, address transla-
tion, accounting, and dial plans. Gatekeepers allow the simple configura- Mandatory and Optional Gatekeeper
tions of the majority of devices on your network, while requiring only a
few devices to hold the majority of configurations. This architecture
Functions
keeps call routing, security, and administration centralized to a few Mandatory Functions
devices. Because the network only has a handful of devices that need to
Admission control: Controls endpoint admission into the H.323
be configured when changes occur, the likelihood of errors is diminished.
network using H.225 Registration, Admission, and Status (RAS)
messages, as follows:
NOTE
I use VoIP in this short cut to simplify the concept. n Admission Request (ARQ)
Gatekeepers will work on any VoXX network using the H.323 n Admission Confirm (ACF)
protocol suite. Gatekeepers do not concern themselves with
the medium or underlying technologies. Gatekeepers only n Admission Reject (ARJ)
need to make path selections and know whether resources are
available for allocation. Admission control is automatically started when the GK is started.
Address translation: Translation of IP addresses to an E.164 phone

number and vice versa.
Gatekeeper Communications
Example 1: 44.44.1.31 = 402-555-5555
The GK operates on three different ports depending on whom it is
communicating with. TCP port 1718 is used when communicating with Example 2: 405-999-9978 = 192.168.1.109. Address translation must
other GKs. TCP port 1719 is used for H.323 devices to register with be configured manually.
GK in their zones. The next port used is TCP port 1720 for call control.

[7]
GATEKEEPERS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Bandwidth control: Provides endpoint bandwidth management Optional Functions

requirements using H.225 RAS messages. When an endpoint decides it Call authorization: The GK can restrict access for endpoints or GWs.
needs a certain amount of bandwidth for the call, it sends a BRQ
message to the GK asking for an amount of bandwidth to be set aside Bandwidth management: Reject admission when the required band-
for the call. The GK responds with a BCF or BRJ message. width is not available. The GK can reject calls if there is not enough
bandwidth to support the call. All active calls are recorded by the GK
n Bandwidth Request (BRQ) to manage the bandwidth within the zone. The GK use the following
n Bandwidth Confirm (BCF) formula to decide whether it can accept or reject the call request.
n Bandwidth Reject (BRJ) Available bandwidth = (Total allocated bandwidth) – (Bandwidth
used locally) – (Bandwidth used by all alternates)
NOTE
Call authorization and bandwidth management must be configured
Messages appear in the order in which they will happen on a manually.
GK-controlled network.
Call management: The GK maintains active call information. Used for
resource allocation / route selection.
Zone management: Manages all registered endpoints in the zone. This
provides centralized control for many devices within the zone. Call management is automatically started. Use show commands to
display this information.
Zone management is automatically started when the GK is started.
Call control signaling: The GK may route call signaling messages
between endpoints using Gatekeeper Routed Call Signaling (GKRCS).
Normally, the GK uses H.225 call signaling. The endpoints send
messages directly to other endpoints without using GK signaling. Cisco
gatekeepers do not support GKRCS.

[8]
H.323 CALL SETUP AND CONTROL CCVP GWGK Quick Reference Sheets by Brion S. Washington
H.323 Call Setup and Control One way to memorize steps of H.323 call setup using a GK:
Now that you understand more about GKs, let’s cover in more depth 1. You get RAS-ed by a 225-lb. bully (H.225), which you have no
how H.323 communicates. control over (UDP) (port 1719).
A. RAS messages are passed through the GK to each device.
Intrazone Call
B. If no GK is present, H.225 messages are sent between
Gatekeeper
Infantry endpoints.
2. You call 931 (Q.931) (H.225) to set up a call with your friends.
V
You do not control them (UDP).
) H.
DP 22
S
(U 5
RA 3. Your 245-lb. friend (H.245) takes control of everyone on the call
RA S
22
5 (U
DP (TCP) (port 1720). Negotiations take place.
H. )
4. Your friends (RTP) over on the negotiated streets and talk to the
bully; you have no control over what happens next (UDP).
H.225 (Q.931) Call Setup (UDP)
Gateway Gateway
Bragg V H.245 Call Control (TCP) V Benning
RTP (UDP)
IP
FIGURE 1.1 Call Control and Setup Using a Gatekeeper

[9]
Steps 2 through 4 are direct between endpoints with no GK Gatekeeper Discovery Process
intervention.
Devices that want to participate on a network must first locate and then
register with a GK. There are two ways to discover a GK. During the
Gatekeeper Routed Call Signaling registration process, it is actually the GWs that register with the GK.
Gatekeeper All devices on an H.323 network are endpoints. Regular endpoints,
Infantry such as phones, faxes, and so on, usually do not register directly to
GKs. Their requests are forwarded by their GWs.
Unicast: The devices use UDP port 1718. Devices are preconfigured
P) with the IP of the GK and can start with the registration process imme-
H H
H
)
P) DP
D
.2 .2
.2
(U
25 4
25
TC (U
AS diately. The GW sends a GRQ to the GK. A GK responds with a

(Q 5 C
R
l( p
tro etu
AS
R
.9 al
25
31 l C
on l S
(U
.2
) C on
gatekeeper confirm (GCF) or gatekeeper rejection (GRJ) message.

l C al
H
al C
P)
al tro
C 1)
lS l
45 93
et (TC
.2 .
up P
H 5 (Q
(U )
Multicast: The devices try to find a GK with the multicast IP

2
D
.2
P)
H
Gateway Gateway
224.0.1.141. The GK responds with GCF or no response at all.
Bragg V RTP (UDP) V Benning
If a GW finds that a GK has gone down, it rejects new calls and
attempts to find a GK; active (established) calls remain operational.
IP Registration Process
1. The endpoint sends a registration request to the GK (port 1718
UDP).
FIGURE 1.2 Example: GKRCS
2. The GK responds with a registration confirm or reject (RCF or
RRJ).
Now that you know the communication process and the protocol used,
let’s discuss how to locate and gain access to the network.

[ 10 ]
Unregistration Process Types of Gatekeepers

1. The endpoint starts the process by sending an unregister request To provide for the different requirements and functionalities needed,
(URQ). GKs are categorized into different types. Each type allows the GK to
2. The GK responds with an unregister confirm (UCF) or unregister handle certain mission-critical tasks. The GK types are active,
reject (URJ). backup/alternate, and directory, as follows:
n Active GK—Controls all H.323 devices within a zone. This is the
Gatekeeper Unregisters Device first GK used on the network. All messages go through this
device; they are then passed on to other GKs as needed.
1. The GK sends a URQ.
n Alternate/backup GK—The second device that handles control
2. The endpoint responds with a UCF or URJ.
of the zone in the event of the failure of the active GK. See the
“Redundancy and Availability” section.
The GK recognizes endpoints by the following:
n Directory GK—Hierarchical GK used to promote scalable growth
n Email IDs of the network. The directory GK holds information on all zones.
n E.164 addresses (telephone numbers) This device helps with Location Request (LRQ) messages. Use of a
n H.323 IDs (user-defined, case sensitive) directory GK helps simplify the process of adding or updating GKs,
because the directory GK knows the location of all zones and GKs.
If you are having registration issues, make sure you have correctly
configured your device with one of them. It is recommended that each When deciding where to put your GKs, you can save yourself a few
endpoint have at least one fully qualified domain name (FQDN) for headaches if you remember a few simple rules:
interzone communication. n Only one active GK per zone.
n Redundancy (backup GK) should always be used in case of
failure.
n Use a directory GK when your network is big or has potential for
large growth.

[ 11 ]
Gatekeeper Topology The basic commands needed to set up a functional intrazone GK are
not complicated. We will go over a basic GK, and then move on to
The GK has a topology that separates areas called zones. Zones are
more complex configurations.
divided into local and remote zones. Local zones are the ones that you
have administrative control over or want to classify as local. Cisco GKs
support up to 100 local zones, but only 1 local zone is mandatory. Basic Intrazone Gatekeeper Configuration
The steps involved in a basic configuration are as follows:
Remote zones are zones that you have no administrative control over,
or they could be zones you have picked to be remote for various other 1. Set up the GK service:
reasons. The use of remote zones is optional (whereas local zones Gk-Infantry # configure terminal
are not). Gk-Infantry (config) # gatekeeper
Gk-Infantry (config-gk) #
Gatekeepers must be aware of their capabilities and where to send calls
they cannot complete. This information is gathered by use of prefixes. 2. Set up the GK local zone:
The two types of prefixes are zone prefix and technology prefix.
Gk-Infantry (config-gk) # zone local GK-Infantry my.army.com
Zone prefixes state which numbers or area codes the device can
complete calls to. The prefix could be the area code for Southern Syntax: zone local gatekeeper-name domain-name
California (714), or it could be an extension in the payroll department 3. Start the GK:
(3367). You can set up your GK to handle or complete any combination Gk-Infantry (config-gk) # no shutdown
of numbers. When planning your dial plan, you must consider many
aspects, from devices used to path selections.
Gk_Infantry running-configuration
Technology prefixes are another way to specify what the GK can and Gk-Infantry# sh run
cannot do. This type of prefix allows devices to be grouped together by Building configuration...
types or services. Examples of types are location, services, and func- Current configuration : 54281 bytes
tions. Cisco GKs use technology prefixes to route calls when there is ! version 12.3
no match for E.164 addresses to a specific GW. hostname Gk-Infantry
! boot system flash:c7200-js-mz.123-4.T7.bin
Gatekeeper
zone local Gk-Infantry my.army.com

[ 12 ]
No shutdown !!— Still under gatekeeper configuration The zone remote command specifies the remote zone, IP address, and
!——Nonrelevant text has been omitted port to use:
After you have a GW or two configured, calls can then take place zone remote other-gatekeeper-name other-domain-name other-gate-
keeper-ip-address [port-number][cost cost-value
across your network. [priority priority-value]] [foreign-domain]
You can use the following options with zone remote:

Interzone Gatekeeper Configurations
n Port-number—Tells what port to use for RAS. Can use between
When you need to make calls outside of your zone, your GW sends your
1 and 65535.
request to your local GK. If the local GK does not know where the remote
endpoint is, it forwards the request to another GK or queries the directory n Cost—Assigns cost to the zone. Can use between 1 and 100;
GK. The zone remote command is used to configure the remote zone. default is 50.
n Priority—Sets the priority of the zone. Value is 1 to 100; default
Gatekeeper Infantry is 50.
Gk-Infantry # configure terminal
n Foreign-domain—Sets the device to a different administrative
Gk-Infantry (config) # gatekeeper
Gk-Infantry (config-gk) # zone local Gk-Infantry my.army.com domain.
Gk-Infantry (config-gk) #zone remote Gk-Calvary my.army.com
192.168.2.1 1719
Gk-Infantry (config-gk) # no shutdown
Gatekeeper Calvary
Gk-Calvary # configure terminal
Gk-Calvary (config) # gatekeeper
Gk-Calvary (config-gk) # zone local Gk-Calvary my.army.com
Gk-Calvary (config-gk) # zone remote Gk-Infantry my.army.com
192.168.2.3 1719
Gk-Calvary (config-gk) # no shutdown

[ 13 ]
Directory Gatekeepers Directory Gatekeeper Call Setup

A Cisco implementation is a device called a directory gatekeeper. The The steps involved in completing a directory GK-assisted call are close
directory GK allows for large VoIP networks without having to main- to interzone, with just a few changes:
tain full meshes between the GKs. When a GK lacks the information to
1. User A dials 5577.
find a remote endpoint, it sends the LRQ to the directory, a lookup
occurs, and the directory forwards the LRQ to the correct GK. 2. GW-1 sends GK Infantry ARQ for permission to make the call.
Directory Gatekeeper Call Setup

3. GK Infantry performs lookup and does not see User B registered.
GK-DIR
GK Infantry performs prefix lookup and finds a wildcard match on
Dir-GK; GK Infantry sends LRQ to GK-DIR and a RIP (Request
V
In Progress) to GW-1.
3 4
4. GK-Dir performs lookup and finds a match for GK Calvary; sends
Gatekeeper Gatekeeper an LRQ to GK Calvary.
Infantry Cavalry
5
V V The remaining steps are steps 4-10 in interzone scenario.

2 3 6 9 8
Directory Gatekeeper Configuration

7
GW-1 GW-2 Directory GKs help create a hierarchical network, which helps promote
V 11 V
controllable growth in the network. This type of design negates the
1
10
need for full-mesh configurations between all the GKs.
The differences between implementing a directory GK instead of intra-

User A User B
Ext 4488 Ext 5577 zone or interzone are minor. The main thing that differs is the number
of commands configured on each device. The GKs are only set up
Zone 1 Zone 2 knowing about their prefixes and where to send unknown prefix
requests. On the other side, the directory GK knows and is configured
FIGURE 1.3 Directory Gatekeeper
with the prefixes for every GK under its influence.

[ 14 ]
Example: Directory Gatekeeper Configuration Gatekeeper Infantry Configuration

zone local Dir_GK myzone 192.168.2.101 zone local Gk_Infantry mycoolzone 192.168.3.202
zone remote Gk_Calvary mycoolzone 192.168.3.201 1719 zone remote Dir_GK myzone 192.168.2.101 1719 foreign-domain
zone remote Gk_Infantry mycoolzone 192.168.3.202 1719 zone prefix Gk-Infantry 714….
zone prefix Gk-Infantry 714…. zone prefix Gk-Infantry 661….
zone prefix Gk-Infantry 661…. zone prefix Gk-Infantry 310…
zone prefix Gk-Infantry 310…. zone prefix Dir_GK *
zone prefix Gk-Calvary 212… lrq forward-queries
zone prefix Gk-Calvary 202… no shutdown
zone prefix Gk-Calvary 312…
lrq forward-queries In the preceding example, calls starting with 714, 661, and 310 are
no shutdown routed through the Infantry GK; calls starting with 212 and 202 are
routed through the Calvary GK.
Gatekeeper Calvary Configuration If a call starting with 714 comes into the Calvary GK, the GK sees that
zone local Gk_Calvary mycoolzone 192.168.3.201
no zone prefix is configured specifically for 714. The GK then uses the
zone remote Dir_GK myzone 192.168.2.101 1719 foreign-domain
zone prefix wildcard * and routes the request to the directory GK. We
can see that the directory GK does have a zone prefix configured for
714 through GK Infantry. The call is then sent to the Infantry GK.
zone prefix Dir_GK * When no E.164 addresses are registered, the GK has two routing
lrq forward-queries
options:
no shutdown
n Route using the technology prefix in the called number.
n The GK assigns default GWs for nonregistered addresses and uses
default GWs matching the technology prefix assigned.

[ 15 ]
Intrazone
Gatekeeper Placement
Now that we have found the GK and understand the call setup process,
Gatekeeper
we’ll look at how and to whom the GK will be providing its services. Infantry
H.323 devices that are within the control of a GK comprise a zone. The
V
zone is not concerned with the network topology; it covers all devices
associated with the GK regardless of location or network segmentation. V V
There is only one active GK per zone. More than one GK can be in a
zone, but the rest are in backup mode.
V V V V
Gatekeepers will operate in an intrazone or interzone network. An
intrazone network is all devices that are under the GK’s control. All the
devices in the zone use the services provided by the GK. Interzone
V V
communications are communications between two or more zones. This
communication process of GKs can be separated into logical and physi-
cal. Logical communication can be one GK controlling multiple zones Zone 1
on the same device. Physical GK communication can involve two or
more GKs, physically separated. For those of you from the data world,
FIGURE 1.4 Intrazone Topology Example, with Both Callers in the Same
think of it as multiple Open Shortest Path First (OSPF) processes
Zone
running on a single router.

[ 16 ]
Interzone
Locating Endpoints
Gatekeeper Gatekeeper
The process of locating endpoints not native to your zone is a fairly
Infantry Cavalry straightforward process:
V V 1. Endpoint 1 sends an ARQ to its GK-A, trying to call another
endpoint.
V V 2. GK-A cannot locate the endpoint and sends an LRQ to GK-B.
3. GK-B responds with a location confirm (LCF) message. The LCF
V V V V
message includes the IP of the remote GW responsible for the
remote endpoint.
4. GK-A responds with an ACF to endpoint 1.
V V
Zone 1 Zone 2
FIGURE 1.5 Interzone Topology Example, with Two Callers in Different

Zones
The difference between intrazone and interzone is the number of GKs

the call must go through and the paths GK messages take. This adds to
the complexity of your network; however, this cannot be helped if you
are calling outside your zone or administrative authority.

[ 17 ]
Interzone Call Setup 1. User A dials 5577.
The steps involved to complete an interzone call are similar in some 2. GW-1 sends an ARQ to GK Infantry for permission to make the
ways to an intrazone call, with a few more steps required to complete call.
the call. 3. GK Infantry performs lookup and does not see User B registered.
A prefix lookup is done, and GK Calvary is selected. GK Infantry
Interzone Call Setup
sends an LRQ to GK Calvary and a RIP (Request In Progress) to
GW-1.
Gatekeeper Gatekeeper
Infantry Cavalry 4. GK Calvary finds User B registered and sends an LCF with
3
GW-2’s IP address.
V 4 V
5. GK Infantry returns with an ACF with the IP address of GW-2.
2 3 5 8 7
6. GW-1 starts a Q.931 call setup to GW-2 User B’s extension.
6
10 7. GW-2 asks permission with an ARQ to accept the call from GW-1.
GW-1 GW-2
V V
11 8. GK Calvary responds with an ACF with GW-1’s IP address.
9
1 9. Plain old telephone service (POTS) call setup.
10. Q.931 connect.
User A User B
Ext 4488 Ext 5577
Zone 1 Zone 2
FIGURE 1.6 Interzone Call Steps

[ 18 ]
Advanced Gatekeeper Configuration n Security acl answerarq—Sets up tokenless call authorization for
devices outside the GK’s zone. This command calls on an access
Options control list (ACL) from matching, allowed IPs.
VoIP networks need to be able to adapt to the constantly changing
demands of the telecommunication world. The needs of your network n Zone subnet—Specifies that discovery and registration messages
might include better resource management, finely-tuned dial plans, or are only accepted from endpoints on the configured subnets. All
securing resources. The world in which we live today has caused other nonconfigured subnets are rejected.
networks to be designed more securely. VoIP security is beyond the Syntax: zone subnet local-gatekeeper-name {default | subnet-
scope of this guide, but I do write about ways to tighten resources; address {/bits-in-mask | mask-address}} enable
which, in turn, can help secure your network a little more. Some people
think that to provide security for a network that a massive security plan
Example: Security Configuration
is needed. This is not necessarily the case in VoIP. Securing a network
Access-list 55 permit ip 99.99.99.1
is a vague term. I will say that securing a network is a multistage plan Access-list 55 remark *** Taylor’s VoIP Network***
that should be implemented as simply as possible. A few commands Access-list 55 permit ip 88.88.88.1
will further your ability to work with a Cisco GK, while serving a dual Access-list 55 remark *** Brittney’s VoIP Network***
purpose of optimization and securing your network.
Gatekeeper
The best way to make sure that your calls are going across as smoothly
Security acl answerarq 55
as possible is to limit unwanted (unknown) endpoints and messages No zone subnet Dir_GK myzone.com default enable
from traversing your network. The following commands provide Zone subnet Dir_GK myzone.com 192.168.2.0 /24 enable
resource-saving techniques and help secure your devices at a basic level.
n Arq reject-unknown-prefix—Forces GKs to reject unknown
prefixes from zones not configured.
n Lrq reject-unknown-prefix—Forces GKs to reject LRQs from
nonconfigured zones.

[ 19 ]
GATEKEEPER TRANSACTION MESSAGES
PROTOCOL (GKTMP) CCVP GWGK Quick Reference Sheets by Brion S. Washington
Gatekeeper Transaction GKTMP Messages

Messages Protocol (GKTMP) All GKTMP messages are either requests or responses. The GK creates
the request, and the external application sends the response.
GKTMP is used for communications between GKs and external appli-
cations where added functions or services are required beyond the Example: Registration Request
capabilities of the GK. GKTMP passes information between Cisco IOS message line
GKs and the external application using ASCII response/request message header 1
messages based on RAS by way of an application programming inter- message header 2
face (API) or over TCP connections. This works in a straightforward message header 3
manner. Some of the incoming requests for the GK are sent to the
external application to fulfill the request. The GK has triggers config- message body line 1
ured for each external application it will use. These triggers are based on message body line 2
message body line 3
RAS tags and values. Incoming RAS messages are checked against the
message body line 4
triggers. If the messages match a trigger, the GK then repackages
the message and forwards it to the correct external application. When the n The message line states what kind of request is being sent.
external application is finished processing, the results are sent to the GK
for further processing, and messages are sent to the requesting endpoint. n The message header is a few lines that contain the external appli-
The main point to remember here is that GKTMP and API are both cation name, GK name, and version ID. Format is field:value.
used to provide this functionality. Your triggers can be manually config- n Blank space is probably used to help with problem resolution.
ured or dynamically configured through API. Static triggers are
n The message body contains the triggers. The message body is
configured via the command line, whereas dynamic triggers are config-
ured via an external application. optional. Format is tag=value.

[ 20 ]
PROTOCOL (GKTMP) CCVP GWGK Quick Reference Sheets by Brion S. Washington
GKTMP Messages Some messages are only sent as a response from the external applica-
tion; they are ACF, ARJ, BCF, BRJ, RCF, and RRJ. The GK sends the
ACF (Admission Confirm)
following messages as requests: DRQ, RAI, and UQ. Also, the applica-
ARJ (Admission Reject) tion server can use a command URQ to send untriggered, unsolicited
ARQ (Admission Request) URQs to unregister an endpoint from a GK.
BCF (Bandwidth Confirm)
BRJ (Bandwidth Reject) Redundancy and Availability
BRQ (Bandwidth Request) Now that we have covered the basic and advanced GK functions, let’s
DRQ (Disengage Request) turn to ways to keep your network resources available for users. Most
networks contain some kind of redundancy and availability to cope
IRR (Information Request)
with network or device failures. Gatekeepers are not immune to hard-
LCF (Location Confirm) ware or software failures, so alternative methods must be engineered.
LRJ (Location Reject) Here are a few ways to help with this task.
LRQ (Location Request)
Gatekeeper Clustering
RIP (Request In Progress)
Gatekeeper clustering provides redundancy for zones in the event of a
RAI (Resource Availability Indication) GK failure. This addition to H.323 allows multiple GKs to control a
RCF (Registration Confirm) single zone. When the GK starts, each cluster member receives a GRQ
from all the other members. The GRQ contains the alternate GK infor-
RRJ (Registration Reject)
mation. The GK opens TCP connections to all cluster members.
RRQ (Registration Request) Clusters can have up to five GKs per clusters. The clusters communi-
URQ (Unregistration Request) cate with Gatekeeper Update Protocol (GUP). When a device registers
with its GK, it is provided with two alternate GKs that can take over
when the primary GK fails.

[ 21 ]
CCVP GWGK Quick Reference Sheets by Brion S. Washington
PROTOCOL (GKTMP)
Gatekeeper Update Protocol n RegistrationIndication—Sent when a connection is made with a

GUP is used to pass information between GKs in a cluster. The infor- new alternate GK, or when a new endpoint registers.
mation included is available and used bandwidth, CPU utilization, GK n UnregistrationIndication—Sent when a GK is down or an
memory, and the number of endpoints registered. The GK’s CPU endpoint is aged out, or when an endpoint unregisters.
utilization and memory are used for resource management. If the GK is
n ResourceIndication—Sent when the GW sends the GK an RAI
overloaded, it asks the endpoint to use another GK with the lowest
message.
resources used. One of the important things to remember about cluster-
ing is that all GKs share registered endpoint information. Therefore, all
GKs within the zone know about all endpoints within the cluster, which Local Zone Clustering
helps with LRQs. zone cluster local cluster-name local-zone-name
GUP Messages Remote Zone Clustering

GUP uses messages to help exchange control and management informa- zone cluster remote cluster name [cost cost-value [priority prior-
tion within the cluster. Here are the messages and when they are used: ity-value]] [foreign-domain][invia inbound-gatekeeper] | [outvia
outbound gatekeeper]
n AnnouncementIndication—Sent every 30 seconds by default, by
all cluster members. The GK updates information on call capacity, Specify devices to participate in with local or remote zone element
CPU load, endpoint load, memory usage, number of active calls, gatekeeper-name ip-address [port].
and number of registered endpoints received from the sending GK.
n AnnouncementReject—Sent if a configuration mismatch exists.
The GK will terminate the GUP connection with the sender.

[ 22 ]
HOT STANDBY ROUTER PROTOCOL CCVP GWGK Quick Reference Sheets by Brion S. Washington
Hot Standby Router Protocol ip address 75.0.0.1 255.0.0.0

standby 2 ip 75.0.0.3 **enables HSRP, creates group 2, virtual
The second method you might think of implementing is Hot Standby IP**
Router Protocol (HSRP). For those of you from the data arena, you are No standby 2 preempt **Not recommended for GKs**
probably already overly familiar with HSRP. HSRP allows devices on standby 2 priority 110 **highest priority will become active,
default is 100**
your network to reach a secondary GW, router, or GK if the primary
standby 2 authentication pepsi **unencrypted authentication
cannot be communicated with. HSRP is invisible to the end user; this is string, must be identical**
achieved by setting up a virtual IP address instead of using the real IP standby 2 timers 5 15
of the device. In the voice world, it can be your GK or GW; in the data !
world, it is your default GW. You must know a few important things !
when implementing HSRP on your voice network. The main point is
that no state information is maintained when failure occurs. Cisco **Infantry config**
recommends using clustering with GKs, which maintains state informa- Hostname Infantry-GK
tion; HSRP is recommend for GWs and directory GKs. Interface Ethernet 0
ip address 75.0.0.2 255.0.0.0
HSRP has different levels of complexity and protection. We will look at Standby 2 ip 75.0.0.3
the minimum commands needed to enable HSRP. Basic HSRP has a Standby preempt
virtual IP address, a virtual MAC address, and a minimum of two Standby authentication pepsi
devices that belong to this newly created group. The primary GK is in Standby 2 timers 5 15
active mode (accepts new requests), and the secondary GK is in !
standby mode (waiting for the active GK to fail). HSRP uses UDP !
port 1985. Hello packets are multicast (224.0.0.2), and Cisco routers
automatically select the virtual MAC address. The devices on your network would be configured to use 75.0.0.3 as
their GK IP.
Example: Basic Two-GK HSRP Configuration
This will provide backup in the event of device failure. You may skip
**Calvary config**
adding the authentication command; this was added to close any secu-
hostname Calvary-GK
rity holes on the GK.
interface Ethernet 0

[ 23 ]
HOT STANDBY ROUTER PROTOCOL CCVP GWGK Quick Reference Sheets by Brion S. Washington
Note n show gatekeeper zone prefix Shows which E.164 prefixes are
routed to which zones.
This example is a simple HSRP configuration. If you need a
more complex configuration or more information, refer to n show gatekeeper zone status Shows zone and configuration
http://www.cisco.com. values.
n show gatekeeper cluster Shows which cluster are configured.
n show gatekeeper performance statistics Shows information on
Verifying Gatekeeper Functions accepted and rejected calls. Will give information based on GK
After going through the lengthy process of setting up your network, it RAS messages. Associated command is clear h323 gatekeeper
is good practice to make sure everything is configured correctly. statistics.
Verification can provide some quality control and assist with resolving
n clear h323 gatekeeper statistics Resets statistics period.
issues. You can use a number of Cisco IOS commands to check your
configurations for typos, errors, or mismatched values. The key to any
verification process is to know which commands to use and then look Problem Resolutions for Gatekeepers
at the output slowly line by line. Here are some of the main commands Even with the most detailed planning, step-by-step implementation, and
to use first: quality checks, problems can arise at any time during the life of your
n show gatekeeper calls Shows active calls GK is aware of. network. I will assume that you are familiar with the Cisco ideology on
troubleshooting. The same philosophy is at play here.
n show gatekeeper endpoints Shows which GWs are registered
with the GK. Is your GK working as expected?
n show gatekeeper gw-type-prefix Shows GWs that have regis- If your GK shows no endpoints registered or your endpoints cannot
tered with a tech prefix and how the GK treats the tech prefixes. locate a GK, you need to check your GK configuration:
n show gatekeeper status Shows GK status (authorization and 1. Make sure the GK service has been defined.
authentication) and zone status.
2. Check your GK and domain names for typos.
3. Issue no shutdown under the GK configuration.

[ 24 ]
VOICE GATEWAYS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Voice Gateways Protocol selection is an important process. It must take into considera-
tion the special needs of the network and how your GWs will function.
The GW is probably one of the most important devices on a VoIP The next major point to consider is the number of options that most of
network. Although the use of a GW is an option on a network, it should the commands offer to further fine-tune the network to the required
be considered mandatory. If you do not use them, you really do not specs. Imagine the protocol available as a pizzeria; there are many
have a real network. pizza places to order from. So, the choice depends on your needs. Does
You can implement GWs with just a few commands on the end devices; the place offer delivery or only take-out? Think of the options as if you
the majority of configurations are located on the GWs. Gateways act as are ordering a pizza. You can choose to add sardines, mushrooms,
the translator between different types of networks, dial planner, security sausage, and anchovies (although you will probably have heartburn
enforcer, and connectivity director to the public switched telephone afterward). This is the same as with your voice network: adding or
network (PSTN) and voice/data networks. Gateways come in analog or changing functions and options add to the heartburn. In the end, you
digital, depending on the requirements of your network. Cisco offers a have your pizza. The moral of this is that to ensure proper functionality,
few different platforms for GWs. Some support only a few connection know exactly what you are adding to your network.
lines; others support many connection lines.
Gateways at the most basic level route calls to the PSTN or network Protocols Used by Gateways
depending on the destination of the call. The other functions GWs Gateways can use a few of the main VoIP protocols on the scene today.
provide include the following: Here are a few that we discuss: H.323, SIP, SCCP, and MGCP.
n Call control and call routing
Complexity with H.323
n Call setup and teardown
H.323 is a complex protocol from its inception. We need to face the
The process of setting up a GW can be either extremely easy or very facts; the protocol was built to accomplish a job. There are less-
complex depending on what functionality the network needs. Your complex protocols (SIP, for instance). H.323 was built to use TCP/IP
configuration can be as little as a few lines for an easy configuration, to and take advantage of its robustness. The components in H.323 are
a few hundred lines for the more complex configurations. All of this terminals, GWs, multipoint control units (MCU), and GKs.
depends on the protocol your company decides to implement.

[ 25 ]
IP IP IP IP
Gateway
Bragg V
NYC Chicago
IP Gateway Gateway
Bragg V V Benning
FIGURE 1.7 Simple Gateway Life
In the preceding diagram, the GW is not connected to another GW IP

or GK.
FIGURE 1.8 GW-to-GW dating
This diagram shows a more complex GW network with a connection to

another GW and a PBX.

[ 26 ]
As you can tell from these diagrams, you can use GWs in a variety of
V network layouts. The only difference is who the GW connects to, how
many devices it is communicating with, and how many commands you
V
need to configure to accomplish it. Keep in mind that these connections
can be over a single interface, multiple interfaces, a single port, or
multiple ports.
V
Session Initiation Protocol
Session Initiation Protocol (SIP) is a simple protocol that uses TCP or
IP
V
UDP. It is built on the client/server model. RFC 2543 defines SIP as a
signaling protocol for creating, modifying, and terminating Internet
V multimedia or Internet telephony sessions.
IP
SIP uses multicast, unicast, or a combination of the two for communi-
IP
cations. SIP can be used by other VoIP protocols to start, locate, and
terminate calls. SIP can interoperate with other VoIP protocols on
devices that allow this. All devices on this type of network are a client,
server, or a proxy. The client (user agent client) sends the invites; the
server (user agent server) provides the service and sends a response.
Proxy servers (user agents) act like both a client and a server; it makes
FIGURE 1.9 The Nightmare requests for others. These requests are handled by the proxy or sent to
another server to fulfill the request.

[ 27 ]
SIP URL addresses follow the user-at-host style (Brion@Wash.com | SIP Call Flow
Brion@1.1.1.5 | 714555@sm4.com). The format is host @ network
address or domain. SIP addresses must be unique. When a SIP client Setup
needs to locate a server, it does so by the server IP and port number in Call Proceeding
INVITE
the Request URI or the locally configured proxy server. Setup

100 Trying
Call Proceeding
SIP can be used for connections to the PSTN, voicemail, and applica-
Alerting
tion servers. SIP offers a variety of services that many companies could 180 Ringing
Alerting
find useful, including the following:
1-Way Voice 2-Way RTP Channel 1-Way Voice
n Fax support
Connect
n Call transfer 200 OK
Connect
n Endpoint locating Connect ACK

ACK
n Conferencing
2-Way Voice 2-Way RTP Channel 2-Way Voice
n Media capability determinations
Disconnect
n Dynamic endpoint registration Disconnect

BYE
n Toll bypass Release

200 OK
Release Complete
Release Complete
FIGURE 1.10 SIP Call Flow

[ 28 ]
SIP Server Call steps using a single SIP redirect server:

SIP
IP
1. The calling party sends an invite to the redirect server.
UAC
2. The redirect server ask the location server to find the path for the
SIP
called party.
RTP
3. The redirect server sends information to the calling party.

4. The calling party sends an acknowledgment of receipt to the
PSTN
V redirect server.
PBX
5. The calling party sends a request to the called party.
FIGURE 1.11 Simple SIP Network 6. The called party responds.
Call steps on a single-proxy-server network: 7. An RTP session is finally opened.
1. The end user registers with a registrar server. SIP Server SIP Server
A-1 B-1
2. The call is initiated by caller. The request is sent via the SIP server
(proxy or redirect). IP IP
3. The server forwards an invite to the called party.

4. The called party responds.
5. The server forwards the called party’s response to the calling
party.
V V
User A User B
FIGURE 1.12 Call Steps to a Remote Network

[ 29 ]
1. User A sends an invite to proxy server A-1. SIP request types:

The location server could also be used to locate User B. n ACK—Confirms the client has received a response to the INVITE
2. Proxy server A-1 contacts proxy server B-1. DNS is used by request.
proxy A-1. n BYE—Terminates a call.
3. Proxy server B-1 contacts user B. n CANCEL—Cancels pending searches; call that is currently in
The location server could also be used here. progress continues.
4. User A and User B trade responses and acknowledgments. n INVITE—Invites to participate in a call.
5. An RTP session is established between User A and User B. n OPTIONS—Queries the capabilities of servers.
n REGISTER—Registers the address with a server.
Connection to a PSTN:
1. All steps are repeated. SIP works by using simple messages to communicate. Here are the
messages:
2. SIP GW and SIP endpoint create RTP session.
n 1xx: Informational—Received request, processing the request.
3. POTS signaling is translated into SIP messages.
n 2xx: Success—The action was successful.
The SIP message format: n 3xx: Redirection—Further action needed.
n A start line n 4xx: Client Error—The request contains bad syntax or cannot be
n One or more header fields done on this server.
n 5xx: Server Error—The server failed to fulfill the request.
n An empty line
n 6xx: Global Failure—The request cannot be fulfilled at any
n A message body (optional)
server.

[ 30 ]
OPEN SET TLEMENT PROTOCOL CCVP GWGK Quick Reference Sheets by Brion S. Washington
Skinny Client Control Protocol Open Settlement Protocol

Skinny Client Control Protocol (SCCP) is a Cisco proprietary VoIP The European Telecommunications Standards Institute (ETSI) Project
protocol used for communications between Cisco CallManagers and IP TIPHON (Telecommunications and Internet Protocol Harmonization
phones. Skinny has a simple architecture; it uses TCP to communicate Over Networks) Open Settlement Protocol (OSP) provides a way for
with the CallManager and UDP between the endpoints. Skinny has a VoIP service providers to use services provided by OSP clearinghouses.
reduced overhead because most of the processing happens on the This client/server protocol is used to exchange accounting, authoriza-
CallManager. IP phones must be configured with the IP address of the tion, and usage information. The purpose of OSP is to manage access
default GW and the IP address of a TFTP server and must have an IP control, accounting, and interdomain routing. You can use this protocol
address for SCCP to work. between H.323 GWs, H.323 GKs, softswitches, and SIP proxies. Public
Key Infrastructure (PKI) is used to provide secure services. The server
Media Gateway Control Protocol creates a secure link for the transfer of information. When the link is
Media Gateway Control Protocol (MGCP) is a simple VoIP text established, a secure token-based signature is used to certify that the
protocol used to control GWs. RFC 2705 defines this protocol as a call is authorized. Clearinghouses provide best route and least cost path
master/slave arch, where call control devices take a specific port on a selections based on subscriber requirements.
GW under control. This protocol offers centralized GW administration.
When centralized administration is used, you can realize large, scalable
VoIP environments. If changes are made, they are made in one area
rather than multiple areas. Therefore, most of your configuration is
done on a few devices while others have bare configurations. Cisco
CallManager uses MGCP on UDP port 2427 to communicate with
GWs. MGCP, like other VoIP protocols, has components that make up
the network. They are endpoints and call agents. Endpoints are voice
ports on GWs controlled by MGCP. The next components are call
agents. Call agents are the external control devices. The call agents
contain all of your dial plans.

[ 31 ]
GET TING A GATEWAY UP AND RUNNING WITH
H.323 CCVP GWGK Quick Reference Sheets by Brion S. Washington
Getting a Gateway Up and Example: Gateway Interface Configuration

interface FastEthernet0/0
Running with H.323 ip address 5.5.5.98 255.255.255.128
h323-gateway voip interface
To set up a functional GW with GK support, follow these steps:
h323-gateway voip id Gk-Infantry ipaddr 75.75.50.1 1719
1. Set up GK and GW registration: h323-gateway voip h323-id GW-1
GW1(config)# gateway
2. Create the H.323 interface: Configuring SIP User Agents

GW1(config-if)# h323-gateway voip interface **under interface Setting up SIP is extremely easy:
configuration mode
1. Enable SIP.
3. Create the H.323 ID:
2. Set up alternate SIP servers (optional).
GW1(config-if)# h323-gateway voip h323-id
3. Set up dial peers.
This command assigns a unique H.323 ID for the device. This ID
is used on the network for identification and connectivity. sip-ua
sip-server ipv4:75.1.1.69
4. Manually configure GK information (optional):
or
GW1(config-if)# h323-gateway voip id (name) ipaddr IP address
port sip-server dns:my-sip.com
This command manually sets up connections to other control

devices (GW or GK). You may leave out this command if you Dial-peer voice 12 voip
need only a basic GW. Session target sip-server

[ 32 ]
ADVANCED GATEWAY OPTIONS MEDIA
RESOURCES CCVP GWGK Quick Reference Sheets by Brion S. Washington
Advanced Gateway Options DSP Codec Functions

Media Resources If the DSP is the heart of the network, it can be said that codecs are the
blood. They encode and decode incoming analog signals into digital
Digital signal processors (DSP) are the heart of any VoIP network. signals and vice versa. There are many codec flavors to choose from.
DSPs provide the most important job of anything on the network. If With so many choices for codecs, there are bound to be tons of options
your network is set up 100 percent correctly, but you have a bad DSP for fine-tuning. Codec configuration can be divided into four group-
located at the beginning or end of a voice path, all is for naught. DSPs ings: global, port, voice card, and dial peer.
are where the encoding and decoding of your voice between analog and
digital devices takes place. All Cisco voice products have some sort of
DSPs; without having this hardware, they would not be able to provide
Voice Card Codec Options
voice support. Because the voice card is where the DSPs are located, this seems like
the logical place to start tuning resources to your requirements.
Cisco has an extensive list of products that can and do provide voice
support. The main components are the voice cards. Voice cards are Codec complexity is a compression technique that states the amount of
what hold the DSP and are made to fit any voice platform. There are processing required for voice compression. The higher the complexity,
quite a few types and configurations of cards. This guide does not the fewer number of calls the DSP can process. When you are using
cover any of the actual cards specifically; it uses general terms and high complexity, your voice channels available will decrease.
hardware. The voice card naming convention will give you an idea of Steps required:
how many calls it can support, and which platform the card will work
in. Let’s go over the resources that DSPs provide to Cisco voice 1. Select the voice card.
networks. 2. Enter the codec complexity you want:
GW-1(config) # voice-card 1
GW-1(config-voicecard) # codec complexity [high | medium | flex] **
Flex is available on C5510 DSP**

[ 33 ]
ADVANCED GATEWAY OPTIONS MEDIA
RESOURCES CCVP GWGK Quick Reference Sheets by Brion S. Washington
The dial peer uses the codec command to specify which codec to use. Assign voice class to dial peer:
The codec command is entered under dial-peer configuration. We go GW-1(config)# dial-peer voice 50 voip
over dial peers in more detail later, so this is just to give you an idea of GW-1(config-dial-peer)# voice-class codec 1
where the codec command fits:
Dial-peer GW-1# Show running-config **all output removed except voice class
and dial peer
dial-peer voice 417001 voip
destination-pattern 417[4,6-9].T
voice class codec 1
codec g723r63
codec preference 1 g729r8
session target ipv4:18.82.16.12
codec preference 2 g729br8
You might also decide to just let the two parties involved negotiate the codec preference 3 g723r63
codec used during the H.245 negotiation process. If this is your prefer-
codec preference 5 g723ar63
ence, you may configure a global class to handle all the negotiations
not specifically configured under individual dial peers.
codec preference 7 g711ulaw
The global command voice class codec enables you to configure the codec preference 8 g711alaw
preference of codec once for any dial peers you want to associate: !
!
GW-1(config)# voice class codec 1 dial-peer voice 50 voip
GW-1(config-class)# codec preference 1 g729r8 description My_house [402]
GW-1(config-class)# codec preference 2 g729br8 destination-pattern 402.T
GW-1(config-class)# codec preference 3 g723r63 voice-class codec 1
GW-1(config-class)# codec preference 4 g723r53 session target ras
GW-1(config-class)# codec preference 5 g723ar63 !
GW-1(config-class)# codec preference 6 g723ar53
GW-1(config-class)# codec preference 7 g711ulaw
GW-1(config-class)# codec preference 8 g711alaw

[ 34 ]
TRANSCODING AND CONFERENCING CCVP GWGK Quick Reference Sheets by Brion S. Washington
Transcoding and Conferencing 4. Configure the voice card for conferencing and transcoding
sessions:
What would the world be like without the ability to talk to multiple
GW-1(config-voicecard)# dsp services dspfarm
people simultaneously? Would the United Nations General Assembly
be able to resolve global problems if there were no translators on staff? 5. Set the number of sessions:
Well, the same can be said about VoIP communications. Media
GW-1(config)# dspfarm confbridge maximum sessions number
resources is the term for DSPs responsible for providing the resources
GW-1(config)# dspfarm transcoder maximum sessions number
to handle conferencing and transcoding. The voice GW uses SCCP to
communicate with the Cisco CallManager, which controls the DSP 6. Enable DSP farming:
farm.
GW-1(config)# dspfarm
7. Set up the resources for conferencing and transcoding on

Configuring Conferencing and Transcoding CallManager.
This section covers how to configure the two likely DSP types you will
find on the exam. The two types are C549, found on Network Modules
(NM), or the C5510, found on the Packet Voice DSP Modules (PVDM).
Setting Up Enhanced Conferencing and
NM-HDV (C549) setup: Transcoding (C5510)
1. Set up the SCCP interface: Enhanced conferencing and transcoding enables you to create multiple
profiles. These profiles allow control of resources and let the GW regis-
GW-1(config)# sccp local interface
ter with multiple CallManager groups.
2. Set up the CallManager IP: 1. Set up the SCCP interface:
GW-1(config)# sccp ccm ipaddr [priority priority-level] GW-1(config)# sccp local interface
3. Enable SCCP: 2. Set up the CallManager IP:

GW-1(config)# sccp GW-1(config)# sccp ccm ipaddr identifier identifier [version ccm
version]

[ 35 ]
TRANSCODING AND CONFERENCING CCVP GWGK Quick Reference Sheets by Brion S. Washington
3. Initialize SCCP: GW-1(config-dsp-profile)# associate application sccp

GW-1(config-dsp-profile)# maximum sessions {hardware | software}
GW-1(config)# sccp
sessions
GW-1(config-dsp-profile)# no shutdown
4. Configure the voice card for conferencing and transcoding
sessions: 8. Associate profiles with CallManager groups:
GW-1(config)# voice-card 0
GW-1(config)# sccp ccm group number
GW-1(config-voicecard)# dsp services dspfarm
GW-1(config-sccp-ccm)# associate ccm identifier priority priority
GW-1(config-sccp-ccm)# associate profile profile register device-
5. Set up the DSP farm profile for transcoding: name
GW-1(config)# dspfarm profile profile transcode **profile is a number range of 1 to 65535
**profile is a number range of 1 to 65535 GW-1(config-sccp-ccm)# bind interface interface
GW-1(config-dsp-profile)#associate application sccp 9. Set up resources in CallManager.

GW-1(config-dsp-profile)#maximum sessions number
GW-1(config-dsp-profile)#no shutdown
6. Set up the DSP farm profile for conferencing:

GW-1(config)# dspfarm profile profile conference
**profile is a number range of 1 to 65535
GW-1(config-dsp-profile)#associate application sccp

GW-1(config-dsp-profile)#maximum sessions number
GW-1(config-dsp-profile)#no shutdown
7. Set up the DSP farm profile for MTP:

GW-1(config)# dspfarm profile profile mtp
**profile is a number range of 1 to 65535

[ 36 ]
DROP AND INSERT MULTIPLEXING CCVP GWGK Quick Reference Sheets by Brion S. Washington
Drop and Insert Multiplexing Example: Drop and Insert for T1

GW-3# show run
The needs of modern businesses have made advances in options that Building configuration...
networks provide. Some come in the form of protocols, others in the
form of hardware. One of these is drop and insert multiplexing. Drop card type t1 0 0 **configures card type
and insert hardware allows companies to buy a single network module. !
These modules support data and voice. They include an integrated no network-clock-participate wic 0 **VWIC won’t use network timing
channel service unit / data service unit (CSU/DSU), and some newer !
models echo cancellation modules. Voice support can be used only for controller t1 0/0
DS0 or PRI modes. Now with a Voice WAN Interface Card (VWIC), framing esf
linecode b8zs
Cisco devices can offer a better value for companies by requiring only
channel-group 1 timeslots 1-12 speed 64 **specify group slots
a single module to be purchased. When they install the VWICs, they
tdm-group 2 timeslots 13-24
can do both data and voice simultaneously. The VWIC can help prevent
!
issues from arising. After the module is in, there is no need to pull the controller t1 0/1
module out to adapt to network changes. framing esf
linecode b8zs
channel-group 1 timeslots 1-12 speed 64
tdm-group 2 timeslots 13-24
!
!
interface Serial0/0:1
ip address 75.75.0.10 255.255.255.0
!
interface Serial0/1:1
ip address 75.75.0.11 255.255.255.0
!
connect t1-xconnect T1 0/0 2 T1 0/1 2 **configures Drop and
Insert between ports

[ 37 ]
ANALOG CONNECTIVIT Y CCVP GWGK Quick Reference Sheets by Brion S. Washington
Analog Connectivity E&M

Since the day the telephone was invented, analog has been the method Most people in the telephony world have heard of Earth and Magnet,
used by phones. Analog waves are converted by the phone into electri- or Ear and Mouth (E&M). E&M is used in analog trunking. These are
cal signal to cross the network; it is converted back to analog at the the connections between central office switches and private branch
destination. Analog has been around for a long time. There are a few exchange (PBX), or for inter-PBX. E&M performs the on-hook and
technologies that have been born in the analog communication world, off-hook signaling to confirm when the handset is picked up or put
as discussed in the following subsections. down. Signaling of the handle’s on-hook and off-hook helps reduce
the chance of two people picking up the phone at the same time.
E&M comes in five variations, as described in the following table.
Analog Signaling
Type Information On-Hook Off-Hook
There are a few choices for analog signaling, depending on what you
1 Used in the M lead connected E lead to ground
require. Signaling is the call control messages sent between devices on United States to ground M lead to battery
the PSTN. Signals include information such as the phone is ringing, E lead open
incoming call, line is busy, and so on. Some of the older signaling
2 Used in the E and M lead open M lead connected
types have been mostly replaced be newer, more efficient signaling. An United States to CO
important part to remember is that analog uses CAS (Channel E lead to
Associated Signaling). CAS sends the control signaling on the same PBX ground
channel as the data. We discuss the types of signaling you can use and 3 Older signaling M lead to CO M lead to CO
how to get them working on Cisco GWs throughout the rest of this ground battery
section. E lead open E lead to ground
4 Not supported N/A N/A
by Cisco
5 Used in the United E and M open M lead to PBX
States and Europe ground
E lead to ground

[ 38 ]
E&M signaling connect PBXs using eight-wire circuits. The break- Setting Up E&M Trunking
down of the wire is four wires for signaling and two or four wires for
E&M ports connect to PBXs, so you must configure the port to match
the voice. Each wire has a different responsibility, as shown in the
the PBX brand you are using:
following table.
Lead Description Pin Number n Call progress tone
SB -48V signaling battery 1 n Impedance
M Signaling input 2 n Operation
R Ring, Audio Input 3
n Signal type
R1 Ring, Audio Input / Output 4
n Type
T1 Tip, Audio Input / Output 5
T Tip, Audio Input 6 Select voice port:
E Signaling Output 7 GW-2(config)# voice-port nm-module/vic-module/port-number
SG Signaling Ground 8 Voice-port 2/1/1
Configure signaling:
GW-2(config-voiceport)# signaling [delay-dial | immediate | wink-
start]
Configure interface type:

GW-2(config-voiceport) # type [1 | 2 | 3 | 5]
** Type 4 is not supported on Cisco devices
Configure country code:

GW-2(config-voiceport) # country country code

[ 39 ]
Configure cabling to use: dial-type Type of out signaling to use for port.
GW-2(config-voiceport) # operation [ 2 wire | 4 wire] dial-type [dtmf | pulse | mf]
Configure PBX termination impedance: ring number Number of rings.

ring number number
GW-2(config-voiceport) # impedance [600c | 600r | 900c | complex1 |
complex2]
signal Type of signaling for port.
FXS and FXO ports
Voice Ports signal [loop-start | ground-start]
The most basic connection type is the port. Without these ports, no supervisory disconnect Enables supervisory disconnect signal.
communications would take place. Ports provide connection to analog
supervisory disconnect
and digital devices. These devices can be a telephone, IP phone, fax,
PBX, and so on. The ports you need to be familiar with are FXO, FXS,
and DID. The FXO and FXS are the ports used in VoIP. The FXS
(Foreign Exchange Station) is what your analog devices are connected FXS Ports
to. FXS ports must supply voltage, ring tone, and power. The FXO FXS ports connect your telephones and fax machines to your GW.
(Foreign Exchange Office) is connected to the PSTN. These ports must provide the same functionality as PBXs. You can
configure a few options for FXS ports, as follows:
busyout Allows busyout of port.
FXO Ports
busyout
FXO ports connect GWs to the PBX. These connections to the port act
like telephones connected to the PBX. The commands used are as cptone Sets call-progress tone for country.
follows: cptone [locale] *** country location
description Sets description for the port.
description Sets description for port.
description [description]
description [description]

[ 40 ]
disconnect-ack Removes line power, and returns acknowledgment DID

of disconnect signal.
It allows calls to bypass an operator. When you call an office, you can
disconnect-ack
either ask the human operator to connect you to the extension, or the
callerid Enables caller ID. PBX can bypass the operator and perform the task itself. The key
points to remember about DID is Cisco offers limited support for DID
callerid
functionality; only few modules support it. The next point is DID
connection Specifies connection type. works on incoming calls only.
connection {plar | tie-line | plar-opx} digits | {trunk digits Configure POTS dial peer:
[answer-mode]}
dial-peer voice 34 pots
ring cadence Sets ring cadence (cptone overrides this command).
Set up direct inward dial:
ring cadence {pattern-number | define pulse interval}
Direct-inward-dial
ring frequency Sets ring frequency.
ring frequency [frequency]
signal Type of signaling for port.

Digital Connections to the PSTN
The voice gateway needs to interface with an assortment of devices
FXS and FXO ports over various connections. Gateways without connections are not of
signal [loop-start | ground-start] much use to a network. The connections can range from small trunks’
DS0 to DS-3. All the connections have the same purpose: to provide a
E&M ports
travel medium and connectivity to other devices. Cisco GWs can be
signal {wink-start | immediate | delay-dial | lmr} outfitted with a variety of modules and cards for connection to almost
any type of network connection. The main connections you must have
station id name Caller ID information sent to caller.
knowledge of and be able to configure are as follows:
station id name[name]
n ISDN PRI
station id number Caller ID information sent to caller.
n ISDN BRI
station id number [number]

[ 41 ]
n Analog 4. Configure the IP address.

n CAS 5. Configure SPIDs (optional).
n Drop and Insert 6. Associate the dial list.
GW-1(config)# isdn switch-type basic-5ess

GW-1(config)# dialer-list 2 protocol ip permit
ISDN GW-1(config)# int bri 0
Integrated Services Digital Network (ISDN) is a digital transport tech- GW-1(config-if)# ip address 5.5.6.12 255.255.255.128
nology used to connect small office, home office (SOHO), or compa- GW-1(config-if)# isdn spid1 74798964631312 8884710
nies to the PSTN. ISDN is a circuit switching technology composed of GW-1(config-if)# isdn spid1 74798964631315 8884714
two types of channels. The Bearer channel (B channel) is the channel GW-1(config-if)# dialer-group 2
that data uses to move across the network. The Data channel (D
channel) is responsible for transporting control protocols. The method PRI Configuration
of sending call control messages over a different channel than the D
1. Configure the ISDN switch type.
channel is known as Common Channel Signaling (CCS). ISDN comes
in two configurations: Primary Rate Interface (PRI) and Basic Rate 2. Configure the controller (T1-E1).
Interface (BRI). PRI is a DS1 that has a capacity of 1.544 Mbps for T1 3. Configure the D channel.
and 2.048 Mbps for E1; BRI has a capacity of between 54 kbps and
128 kbps. GW-1(config)# isdn switch-type primary-5ess
GW-1(config)# controller T1 1
GW-1(config-controller)# framing esf
BRI Configuration
GW-1(config-controller)# clock source line secondary 1
1. Configure the ISDN switch type. GW-1(config-controller)# linecode b8zs
2. Create a dial map. GW-1(config-controller)# pri-group timeslots 1-24
3. Configure the BRI.

[ 42 ]
D-Channel Configuration CAS Configuration

GW-1(config)# interface Serial1:23
Channel associated signaling is sent over the same channels as the
GW-1(config-if)# no ip address
data/voice. It is used primarily with PBXs over E1 or T1 lines. CAS is
GW-1(config-if)# isdn switch-type primary-5ess
also known as robbed-bit signaling, because some of the bits used to
GW-1(config-if)# isdn incoming-voice modem
carry data information are robbed for signaling purposes. Although
GW-1(config-if)# isdn send-alerting
some bits are stolen for signaling, all channels are still available for
Now that we have ISDN BRI and PRI set up, we turn our attention to carrying the data.
verification and troubleshooting. You can use some good commands to GW-5(config)# controller T1 0
help verify your configuration, or to resolve issues that pop up. One GW-5(config-controller)# framing esf
extremely vital area of ISDN is the cause codes. ISDN cause codes are GW-5(config-controller)# clock source line primary
present in numerous voice and ISDN command outputs. You can find GW-5(config-controller)# linecode b8zs
an extensive list of ISDN cause codes at www.cisco.com. GW-5(config-controller)#ds0-group 0 timeslots 1-24 type fxo-ground-
start
The show isdn status command displays Layer 1 through Layer 3
information. If status is not active, you need to check your connection. Syntax: ds0-group ds0-group-number timeslots timeslot-list type
Also, check to make sure MULTIPLE_FRAME_ESTABLISHED is {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-
present. start | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-
start}

[ 43 ]
CIRCUIT EMUL ATION CCVP GWGK Quick Reference Sheets by Brion S. Washington
Circuit Emulation or
GW-5 (config-controller)# framing {crc4 | no-crc4 | unframed} **E1
Circuit Emulation over IP (CEoIP) is an end-to-end circuit emulation
framing
service for time-sensitive data over an IP network. This transparent
GW-5 (config-controller)# clock source {internal | line | adaptive
transport is used by applications that need a constant bit rate. The channel-number}
endpoints can use adaptive clocking to sync the clocks. CEoIP has a GW-5 (config-controller)# cablelength {long | short} {attenuation |
few optional features, such as CAS, data protection, and payload length}
compression. GW-5 (config-controller)# crc-threshold value
GW-5 (config-controller)# description text
When and why you should use CEoIP: GW-5 (config-controller)# loopback{local {line | payload} | network}
n Automated industrial environments Creating CEM channels on the T1/E1 line:

n Crypto tunneling GW-5 (config-controller)# cem-group group-number {unframed |
n Legacy services and applications timeslots timeslot
n Radar data [speed {56 | 64}]}
n Satellite data Configuring the connection using xconnect:

GW-5(config)# cem slot/port/channel
GW-5(config-cem)# xconnect remote-ip-address virtual-connect-ID
Configuring CEoIP encapsulation
Configuring the NM-CEM-4TE1 card: encapsulation-type
GW-5(config-cem-xconnect)# local ip address ip-address
GW-5# configure terminal GW-5(config-cem-xconnect)# local udp port port-number
GW-5(config) # card type {t1 | e1} slot GW-5(config-cem-xconnect)# remote udp port port-number
Configuring the T1 /E1:

GW-5 (config) # controller {t1 | e1} slot/port
GW-5 (config-controller)# framing {esf | sf | unframed} **T1
framing

[ 44 ]
FA XING CCVP GWGK Quick Reference Sheets by Brion S. Washington
Faxing Modem Pass-Through

The same reasons for running voice over data are the same reasons for This method is like fax pass-through except there are modems at both
faxing over data. Although faxing probably does not take as long to ends of the connection. Modem pass-through uses the same transparent
complete as a phone conversation, expenses can be high for long- connection, and G.711, with no VAD. The supported protocols are
distance faxing. There are a few ways to get that important fax across a H.323, MGCP, and SIP.
voice network. Each of them was developed for specific situations.
Modem Relay
Cisco Fax Relay Modem relay is a method built to deal with overhead. It uses Simple
This Cisco proprietary protocol was created to send faxes over a VoXX Packet Relay Transport (SPRT) to transport modem signals over IP. The
network. The DSP acts like a virtual fax machine, converts the signal SPRT GW converts analog to digital. SPRT uses the UDP protocol, and
back to its original form, and then sends it to the remote fax machine. is less sensitive to jitter and clocking issues than modem pass-through.
To save bandwidth, it uses 9.6 kbps. A key point to know is why a fax The supported protocols are H.323, MGCP, and SIP.
cannot be treated like a normal call. When a fax comes into the GW, it
cannot be treated like a normal voice call, because otherwise the GW
will miss the fax signaling and continue to treat the call as a normal T.37 Fax Store and Forward
voice call. T.37 is the way to send faxes across your network like email attach-
ments. The document is scanned, converted to TIFF, and then sent to a
receiver’s email address using Simple Mail Transport Protocol (SMTP).
Fax Pass-Through The Cisco way of implementing T.37 is by way of GWs. On-ramp
This method uses a transparent IP connection to transport the fax. In GWs do the conversions and sending of faxes; the off-ramp GWs are
this mode, no conversions take place. It treats the fax like a voice call where the messages are received or held until forwarded. T.37 uses
with no special fax capabilities. The codec used is G.711, with no Multi-Media over IP (MMoIP) dial peers. Interactive Voice Response
Voice Activity Detection (VAD), and supports H.323, MGCP, and SIP. (IVR) applications are used for on-ramp and off-ramp functions.

[ 45 ]
TCL SCRIPTS CCVP GWGK Quick Reference Sheets by Brion S. Washington
T.38 Fax Relay You can accomplish a lot of things when you have access to a scripting
language. On the GWGK exam, you’ll be able to narrow down the
Like all things in life, there is an industry standard, and then there is
extensive list of things you’re able to do and concentrate on only a few
the Cisco way. Well, T.38 is the industry standard for fax relay. The two
things. What you need to do for your exam is as follows:
ways are similar, but T.38 needs to have a T.38 GW on both ends of the
connection. The main difference is that T.38 allows faxing to virtual fax n Download Tcl scripts.
machines compatible with T.38. This protocol uses VoIP dial peers. The
n Configure Auto Attendant.
supported protocols are H.323, MGCP, and SIP.
n Describe usage (billing, radius, AAA).
Tcl Scripts
Tool Command Language (Tcl) is an interpreted scripting language. Download Tcl
This means that you do not need to compile anything to use it. The Downloading the Tcl software is fairly simple. You need to make sure
language provides a set of commands that give the basic functions. If your Cisco IOS release supports this functionality. Next, you need a
more advanced or specialized functions are required, you can use CCO login to download Tcl from http://www.cisco.com:
extension sets to augment commands. Cisco has created an extension
set for IVR functionality. A Tcl IVR script is activated when a call http://www.cisco.com/cgi-bin/tablebuild.pl/tclware
comes into the GW. http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp

[ 46 ]
TCL SCRIPTS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Get scripts from a TFTP server: 3. Choose a name and pilot number to use by every AA application.
archive tar /xtract tftp://75.75.10.240/cme-mytcl.tar flash: 4. Choose the number and type of call routing (menu) options.
5. Customize your prompts.
6. Set up call-retry parameters.
Auto Attendant
Auto Attendant is configuring your CallManager or CallManager 7. Choose where to send unanswered calls or full-queue calls.
Express to auto-answer incoming calls. When a call comes in, the Auto
Attendant can provide a greeting, followed by prompts to help direct
Download the Audio Files and Tcl Scripts
the call to the right group. You can also use Auto Attendant to provide
information and help to begin problem resolution. You configure Auto 1. Download files to a TFTP server accessible by CallManager.
Attendant in a three-step process: 2. Uncompress files in privilege mode:
1. Plan the call flow. GW-4# archive tar /xtract tftp://75.75.100.240/cme-

yourscript.tar flash:
2. Download the audio files and Tcl scripts.
3. Configure incoming dial peers.
Configure Dial Peers

Call Flow Design dial-peer voice 44 pots
service my_cool_script
You must determine how to handle incoming calls. When calls come in,
incoming called-number 40255511
how many queues do you have? You must make routing decisions. You
port 1/1/0
must consider options to offer callers.
1. Choose a name for the call-queue service.
2. Decide how many authentication, authorization (AA) applications
are needed.

[ 47 ]
IP-TO-IP GATEWAY (IPIPGW ) CCVP GWGK Quick Reference Sheets by Brion S. Washington
IP-to-IP Gateway (IPIPGW) IPIPGWs are used like other voice GWs in providing interprotocol
networking. With interprotocol networking, you translate one type of
The main points you need to understand about IPIPGWs are the serv- VoIP protocol used by the calling party into a different protocol used
ices they provide for your network and the key reasons they are not by the called party. IPIPGWs work with H.323 or SIP and require
considered normal voice GWs. special Cisco IOS (IPIPGW Image) software to perform these func-
IPIPGWs offer services such as video support, interprotocol network- tions. Another functionality of these types of GWs is the capability to
ing, address hiding, and better billing capabilities. support video calls. Video codecs supported are Far-End Camera
Control (FECC), H.261, H.263, and H.264.
Billing can be simplified with the addition of IPIPGWs. Sometimes
you need to bypass the PSTN and send VoIP between two VoIP The following figure shows an example and the likely steps used to
networks. There are many names used to refer to this type of device: provide IPIPGW in a real-world VoIP network.
IP-to-IP gateway, border element, and session border controller.
IPIPGW
Wholesale VoIP resellers (WVR) or Internet telephony service
Company A
providers (ITSP) use this method to connect to other VoIP resellers.
One of the main reasons for implementing this kind of connection is
V
for billing. When a WVR terminates calls between two other WVRs,
the middleman misses out on billing opportunities because it does not Chicago
control the terminating PSTN GW. The originating and terminating

GWs see the call setup and release messages; the GWs in the middle V V
do not. Both WVRs and ITSPs basically do the same thing, kind of like Company Z Company D
“tomayto or tomahto.” It’s all in how you perceive them.
FIGURE 1.13 WVR Example
IPIPGW can hide IP addresses of your service providers from
customers, making your network look larger than it actually is. Address 1. Company A is a small company of five employees. They would
hiding can be a useful tool to expand your company’s size without like to enter the wholesale VoIP / ITSP market. Company A buys a
actually buying more equipment. The reason I say this is your company Cisco MultiService Gateway.
can now offer more destinations to clients without the fear of clients 2. Company A buys a route from Company D to Chicago.
acquiring the destinations from your service provider. This type of
3. Company A is assigned a tech prefix of 333313# by Company D.
address hiding is widely used by wholesale VoIP companies.

[ 48 ]
IP-TO-IP GATEWAY (IPIPGW ) CCVP GWGK Quick Reference Sheets by Brion S. Washington
4. Company A sells VoIP minutes to Company Z for calls to Chicago. IPIPGW H.323 or SIP Setup
5. Company Z is assigned a tech prefix of 55# by Company A. 1. Enter voice services configuration:
6. Company Z has end users (callers) with family calling the Windy GW-1(config)# voice services voip
City. Calls are sent to Company A’s IPIPGW IP address with a
tech prefix of 55#. 2. Specify allowed connections:
GW-1(config-voi-serv)# allow connections [h323 | sip] to [h323 |
7. Company A’s IPIPGW strips off 55# and forwards digits with sip]
333313# appended to Company D’s IPIPGW.
8. Call detail records (CDR) are checked on all minutes from By default, H.323-to-H.323 connections are disabled. Any-to-POTS
Company A’s IPIPGW for billing. and POTS-to-any are enabled by default.
You can repeat Steps 2 through 8 as many times as necessary for each
new route or company you add. The only thing that will stay constant is
the tech prefixes assigned. The tech prefixes are assigned only once,
when the first route is added, and remain the same throughout the busi-
ness relationship.
So, Company A now looks like a big WVR or ITSP, without the need
to have equipment located in every destination they want to offer. There
are advantages and disadvantages to this style of business.
Advantages Disadvantages
Centralized billing No control of remote networks
Centralized security Limited quality of service support
Low investment to start up Test every new route
Unlimited growth potential Constantly changing providers

[ 49 ]
ADVANCED IPIPGW COMMANDS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Advanced IPIPGW Commands voice class codec 1

Like any other technology, you can get up and running with some basic codec preference 2 g729br8
configurations, but you always have other options to help tweak your codec preference 3 g723r63
network. Following here are some of the options available for IPIPGW. codec preference 4 g723r53
Codec transparency is used to pass codec capabilities between codec preference 6 g723ar53
endpoints. This command is the same as for normal GWs. codec preference 7 g711ulaw
codec preference 8 g711alaw
1. Configure the voice class.
!
2. Set the preferences of codecs. !
dial-peer voice 50 voip
3. Assign the voice class to dial peers.
description My_house [402]
destination-pattern 402.T
GW-1(config)# voice class codec 1
voice-class codec 1
GW-1(config-class)# codec preference 1 g729r8
session target ras
GW-1(config-class)# codec preference 2 g729br8
!
You can use the same scenario with a GK. When a GK is used, the only
differences are new commands and terminology. Via zones are zones
GW-1(config-class)# codec preference 7 g711ulaw
that have IPIPGW and via zone-enabled GKs. Via zone-enabled GKs
GW-1(config-class)# codec preference 8 g711alaw
send transient traffic to the IPIPGW. The IPIPGW terminates the call
and then re-originates the call to the destination. Because the middle-
Assign the voice class to dial peers: man can now see the call setup and call release, calls that normally
would have been unbilled or not allowed are completed and billed.
GW-1(config)# dial-peer voice 50 voip
GW-1(config-dial-peer)# voice-class codec 1

[ 50 ]
ADVANCED IPIPGW COMMANDS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Company A
Gatekeeper Via Zone Gatekeeper Setup
V
Local zone:
V
Chicago 1. Enter the GK configuration:

V V
Company Z Company D 2. Configure the local zone:

Gk-Calvary(config-gk)# zone local cope mydomain.com 75.75.86.97
FIGURE 1.14 Via Zone Gatekeeper Example 11719 invia Gk-Ranger outvia Gk-Ranger
zone remote zone-name domain-name ip-address [port number][invia
gatekeeper name | outvia gatekeeper name
Note
This example is a basic via zone topology. I did not place the Remote zone:
remote GKs in here to help simplify the theory.
1. Enter the GK configuration:
2. Configure the remote zone:

Gk-Calvary (config-gk)# zone remote pepsi mydomain.com 75.75.86.91
11719 invia Gk-Rebel outvia Gk-Rebel
zone remote zone-name domain-name ip-address [port number][invia

gatekeeper name | outvia gatekeeper name

[ 51 ]
GATEWAY REDUNDANCY CCVP GWGK Quick Reference Sheets by Brion S. Washington
Advanced Gateways Options Gateway Redundancy

The ability to fine-tune the network can lead to better resource usage, Gateways also need to be configured with redundancy in case the
help reduce chances of errors, and aid in problem solving. You can primary GK or CallManager fails to diminish issues from arising. We
configure literally hundreds of commands for a GW. You do not need to cover three ways to accomplish this: Alternate GK, HSRP, and
know all of them for the test, but you do need to know them for real- Redundant CallManager.
world applications. I will go over a few of the commands I believe are
important for the GWGK exam:
GW1(config)# Voice service [voip | voatm | pots | vofr] **used to
Alternate Gatekeepers
apply voice services globally*** A GW can be configured with up to two alternate GKs:
GW1(config-voi-serv)# Allow connections from-type [any | h323 | sip Interface FastEthernet0/0
| pots] to to-type [any | h323 | sip | pots] ** this option under Ip address 5.5.5.98 255.255.255.128
voice service command decides which VoIP connections the GW will h323-gateway voip interface
allow. Use the No form to disallow. h323-gateway voip id Gk-Infantry ipaddr 75.75.50.1 1719
h323-gateway voip id Gk-Calvary ipaddr 75.75.66.1 1719
h323-gateway voip h323-id GW-1
Note
Alternate GK is here because the configuration is done on the
GW and not the GK. It can be argued that it can also go in
the GK section.

[ 52 ]
SUR VIVABLE REMOTE SITE TELEPHONY CCVP GWGK Quick Reference Sheets by Brion S. Washington
Redundancy to CallManager Clusters Survivable Remote Site

You must configure a dial peer for each CallManager in the cluster you
want to use. Remember that the preference command is important to
Telephony
the order in which they are selected. If you were to design a modern VoIP network, your client would proba-
bly ask for a voicemail system, connections to the PSTN, remote sites,
Dial-peer voice 199 voip and so on. The first question you would be asked as the designer is
Session target ipv4:192.168.1.11
what happens if we lose our network? How will we make and receive
Preference 1
calls? Your response can be this: Because a Cisco solution was
installed, there is no problem in the event of network failure.
Dial-peer voice 200 voip
Session target ipv4:192.168.1.12
Survivable Remote Site Telephony (SRST) provides Cisco IP phones a
Preference 2
backup path in the event of network failure, but only if they are
connected to a local Cisco device. SRST makes sure your employees
Dial-peer voice 201 voip have at least basic phone service. When IP phones lose connectivity to
Session target ipv4:192.168.1.3 all CallManagers, SRST uses the GW until connectivity to a CM can
Preference 3 be reestablished. Your phones will kick into SRST in about two to three
minutes. The phones must wait three times the keepalive period and try
You need to repeat this for all CallManagers. The preference range is 0 all your other CallManagers. In the event SRST failover occurs, your IP
to 10; 0 is the default. phones will periodically check whether a Cisco CallManager is avail-
able starting after two minutes of being in SRST mode. When commu-
nications have been reestablished with a Cisco CallManager, all phone
service returns to normal. Here is what it looks like.

[ 53 ]
SUR VIVABLE REMOTE SITE TELEPHONY CCVP GWGK Quick Reference Sheets by Brion S. Washington
to make sure that you’re running DHCP off your GW and not the
PSTN
CallManager. If your addresses are assigned by the CallManager, you
CM 1 will lose your lease. SIP requires that the GW act as a registrar server
during SRST. It is also recommended that for faster failover, SIP
IP IP
should register with both the proxy and registrar server.
SRST can be set up and will work if your network meets the following
criteria:
SRST Router
V V
n Cisco IP phones are connected to a Cisco device capable of SRST.
n MGCP Fallback and SRST are both configured (MGCP only).
n H.323 and SIP are getting call control and dial plans from the GW.
CM 2
DHCP commands for the SRST router:
IP
ip dhcp pool pool-name **Creates DHCP pool to assign IPs from**
Central Office Remote Office
network ip-address mask prefix-length **IP of DHCP pool**
option 150 ip ip-address **IP of TFTP server to download image**
FIGURE 1.15 SRST default-router ip-address **IP of router phones are connected to**
When SRST is configured, different protocols respond differently to the SRST for H.323:
loss of Cisco CallManager connectivity. You should pick the protocol GW1(config)# call-manager-fallback
that best fits your survivability needs. Each of the protocols has special GW1(config-cm-fallback)# ip source address 192.168.1.90
requirements that you must address. An MGCP has its own form of GW1(config-cm-fallback)# max-ephones 10 ** states how many calls
SRST called MGCP Fallback. You can configure both of these features will be allowed. This value depends on the Cisco IOS version and
platform you are running.
on the same device in later Cisco IOS releases. H.323 will not keep
GW1(config-cm-fallback)#max-dn 8 **states the number of directory
active calls up if there is no signal from a Cisco CallManager. The numbers supported. This is also platform dependent. Check command
unofficial workaround to this is no h225 timeout keepalive. This reference guide for some max-dn.
might help to keep calls active in the event of a link failure. You need

[ 54 ]
GATEWAY PROBLEM SOLVING CCVP GWGK Quick Reference Sheets by Brion S. Washington
SIP SRST SRST on MGCP Gateways

When SIP goes into SRST mode, phones registered with a proxy server MGCP requires that both SRST and MGCP Fallback be configured. In
can use the voice GW acting like a SIP registrar. addition, you must also create a dial peer with basic PSTN connectivity.
1. Specify allowed connections. GW1(config-)# ccm-manager fallback-mgcp
GW1(config)# service [alternate | default] service-name location
2. Set up the GW as registrar.
GW1(config)# voice service voip GW1(config)# exit
GW1(config-voi-serv)# allow-connections h323 to sip
GW1(config-voi-serv)# allow-connections sip to h323
GW1(config-voi-serv)# allow-connections sip to sip
GW1(config-voi-serv)# sip Gateway Problem Solving
GW1(config-serv-sip)# registrar server Verifying and resolving issues with GWs can be accomplished with
some of the commands you are already familiar with. You can verify
which GKs are configured by one of two commands.
H.323 SRST Reminders show interface Show the VoIP interface and its configuration.
1. You need to configure POTS dial peers to allow direct extension. show gateway Shows the list of GKs configured, along with their
2. Auto Attendant is provided by Tcl scripts. priority values.
3. Voicemail during SRST must be planned. There are a few ways of The last few places to check are H.323 configurations on the GW. You
doing this. will need to check the interface and that your GK information was
entered correctly. Next, you can make sure that you have general
There are more things that must be considered for phone functionality network connectivity.
during SRST. I picked these because of their high daily usage.

[ 55 ]
DIAL PEERS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Dial Plans dial-peer voice (#) MMoIP
Dial plans are an essential part of completing a call. They tell the GWs dial-peer voice (#) VoFR
where to send the call and how to treat the digits dialed. Dial plans are dial-peer voice (#) VoATM
created using a combination of commands to perform call routing. The no shutdown
main components of dial plans are dial peers and their subcommands.
Dial peers are probably the most important aspect of your network The number range (#) is from 1 to 2147483647. Each dial peer must
configuration. They enable you to do almost anything you could think have a unique number assigned to it. If you assign the same number to
of to do with incoming or outgoing calls. The responsibility the dial dial peers, the last configured dial peer overwrites/changes dial-peer
plan carries means your dial plan should undergo an extensive design parameters.
and planning process to fulfill all requirements and functions. Dial
plans can be located on GWs, CallManagers, PBXs, and softswitches. MMoIP dial peers are used mainly for fax capabilities. SMTP messages
are forwarded to a server.
VoIP dial peers are used for connections on packet voice networks. Dial
Dial Peers strings are mapped to an IP address or remote IP.
Dial peers come in a few types: MMoIP, POTS, Voice over ATM
(VoATM), Voice over Frame Relay (VoFR), and VoIP. They allow the POTS dial peers are used to send calls to telephony devices or to the
GW to provide connectivity to the different network topologies, the PSTN. Dial strings are mapped to voice ports.
PSTN, or mail servers. Most VoIP protocols can use dial peers. They VoFR dial peers define connections that will use FRF.11 on a Frame
are H.323, SCCP, and SIP. The basic configuration on these dial peers Relay network. Dial strings are mapped to a permanent virtual circuit
are similar; they vary in only a few commands and wording of the (PVC) or a switched virtual circuit (SVC).
commands. Dial peers are picked by the longest matching number
dialed. All dial peers must be in the UP operational state to be consid- VoATM dial peers are used with AAL5 over ATM networks. Dial
ered for matching. strings are mapped to a virtual circuit (VC).
Dial-peer syntax and types: Note

dial-peer voice (#) VoIP Not all Cisco platforms support all five types of dial peers.
dial-peer voice (#) POTS

[ 56 ]
CALL LEGS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Valid and Invalid Dial Peers Dial Peer Matching

All dial peers must be in operational state to be considered for match- Understanding how dial peers are matched to call legs is essential to a
ing. When designing your dial peers, you must know which subcom- properly functioning voice network. Let’s go over how a GW matches
mands can be used together without adversely affecting the dial peer. inbound and outbound dial peers. It is assumed that you are familiar
There are three combinations that can be used to place peers in opera- with how telephony works, so no in-depth explanation is given.
tional state:
Inbound calls use the entire string and are checked against configured
n An answer-address is configured. dial peers in the following order:
n Destination-pattern with session-target or a voice-port. 1. incoming-called-number (called number “DNIS” information
n Incoming-called-number configured. used).
2. answer-address (calling number “ANI” information used).
To make sure all your dial peers are in the operational state you want
3. destination-pattern (calling number “ANI” information used).
them in, enter show dial-peer voice summary command.
4. voice-port (that has dial peer port configured).
Call Legs Note
Calls processed by the GW are categorized in one of two groups: AS5xxx series GWs stop after Step 3. Dial peer 0 will be
used, and the call will be treated like a modem call. You
inbound call leg or outbound call leg. By knowing how the GW sees might want to store this tidbit in your memory for possible
the call, you will be able to set up your dial peers according to the future problem solving.
network requirements.
All calls have two call legs, the inbound and outbound legs. Inbound Outbound dial peer matching is much easier to remember. The choices
legs are the calls coming into the device; outbound legs are calls sent are limited to only two options. All outbound dial peers are matched on
from the GW. These two legs are used to match which dial peer to use the destination-pattern. The POTS peer then sends the call out the
for completion of the call. With the flexibility to match either call leg, configured port; the VoIP peer sends the call to the value specified in
the dial plan should be able to handle most dial scenarios with ease. the session-target.

[ 57 ]
CALL LEGS CCVP GWGK Quick Reference Sheets by Brion S. Washington
Default Dial Peers destination-pattern command

While we are on the subject of dial peer selection, let’s look at default This command can decide which dial peer to use and the destination.
dial peers. When no matches can be found, the GW has two choices: The variable can be an E.164 or user-defined number. The string can
drop the call or try to complete the call any way that it can. That is consist of 0 through 9, A through D, and a few special characters.
where the default peers come into practice.
Special characters:
Default POTS dial peer configuration:
n The asterisk (*) and pound sign (#) are on touchtone dial pads.
n No IVR applications n Backslash symbol (\), followed by a single character, matches the
character.
Default VoIP dial peer configuration:
n Brackets ([ ]). A range is a sequence of characters in the brackets;
n Any codec only 0 to 9 are allowed.
n Fax-rate voice n Circumflex (ˆ), a match to the beginning of the string.
n IP precedence 0 n Comma (,), places a pause between digits.
n No Resource Reservation Protocol (RSVP) n Dollar sign ($). Null string at the end of the input string.
n VAD enabled n Parentheses (( )). A pattern that is the same as the regular expres-
sion rule.
An easy way to remember this is that the default values can be used on
n Percent sign (%). Preceding digit occurred zero or more times;
default dial peers, and you must sacrifice all the bells and whistles to
similar to wildcard usage.
complete a generic call.
n Period (.). Wildcard matches any digits entered.
n Plus sign (+). Preceding digit occurred one or more times.
n Question mark (?). Preceding digit occurred zero or one time.
n (T), variable digit string. Router will wait until all digits are
entered.

[ 58 ]
DIGIT MANIPUL ATION CCVP GWGK Quick Reference Sheets by Brion S. Washington
answer-address command Hunt Groups

Identifies the incoming dial peer associated with the VoIP call. POTS Hunt groups are used to add redundancy or routing choices to dial
calls are associated with the port the call came in on: plans. They work by grouping dial peers that can complete specific
n The asterisk (*) and pound sign (#) are on touchtone dial pads. calls into a hunt group.
n Comma (,) places a pause between digits. The commands to configure hunt groups and assign dial peers are as
follows:
n Period (.). Wildcard matches any digits entered
hunt group (configured in global configuration mode).
n (T), variable digit string. Router will wait until all digits are
entered. hunt group (configured under each dial peer you want to assign to a
hunt group).
huntstop (stops dial peer hunting if call fails).
dtmf-relay command
How H.323 and SIP GWs relay dual-tone multifrequency (DTMF)
tones between the telephony interface and the IP network. Digit Manipulation
Syntax: dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal] Digit manipulation is something that you will definitely use at some
[rtp-nte] [sip-notify] point in your network. The idea behind this is pretty straightforward.
Your GW receives digits from an outside source, and you want to do
one of the following to the received digits:
progress-ind command n Change digits
This command allows progress indicators to be sent during a call to
n Add digits
help with signaling.
n Subtract digits
Syntax: progress_ind {setup | connect | progress | alert} {enable pi-
number | disable}

[ 59 ]
VOICE TRANSL ATION RULES AND PROFILES CCVP GWGK Quick Reference Sheets by Brion S. Washington
These are the main ways to manipulate digits. We manipulate digits for
various reasons (for example, virtual private networks [VPNs], least
Voice Translation Rules and
cost routing, path selection, billing, and accounting). Manipulation of Profiles
digits can make calling easier for the end users. Users can enter a few Voice translation rules and profiles provide a powerful way to translate
digits, and the GW can change the digits according to administrator digits. The translation rules allows the called number and dialed
wishes, but will add to the complexity of the configurations. That is the number identification services (DNIS) digits to be manipulated. Voice
trade-off you must make. translation profiles use voice translation rules and can change every-
thing the rules can, plus they allow the redirecting number to be
changed.
Commands for Digits Manipulation
n digit-strip
Used only on POTS dial peers only. When a call comes in on a POTS
Voice Translation Rule
dial peer, the matched numbers are stripped, and the remaining Wildcard patterns are used to help augment the pattern matching.
numbers are forwarded on. Configured under POTS dial peer, this . Any single digit.
command is enabled by default.
0 to 9, *, # Any specific character.
Usage: [0-9] Any range or sequence of characters.
* Match none or more occurrences.
digit-strip
+ Match one or more occurrences.
n num-exp ? Match none or one occurrence.
Sometimes companies want to make dialing numbers easier for
customers or employees. The num-exp command can take dialed digits Syntax: rule precedence /match pattern/ /replacement pattern/
and translate them into a different length of digits.
Example:
Examples: voice translation-rule 1
num-exp 402…… 1402…… **1 is added to beginning rule 1 /^9...#809/ /809/
num-exp 17142638080 8080 **incoming number changed to 8080

[ 60 ]
VOICE TRANSL ATION RULES AND PROFILES CCVP GWGK Quick Reference Sheets by Brion S. Washington
rule 2 /^9...#1809/ /809/ 3. Apply profile:

rule 3 /^35681809/ /809/
n Globally
rule 4 /^9...#1/ /1/
rule 5 /^35681876/ /876/ n Dial peer
rule 6 /^3568011/ //
n Interface
rule 7 /^9...#/ //
Cisco offers a test command to see whether your rule is working n Source IP group
correctly: n Trunk group
test voice translation-rule name-tag input-number
n Voice port
Global example:
Voice Translation Profile voice translation-profile VOIP-out
When you use this, you need to follow three steps: translate called 1
1. Create at least one voice translation rule. Dial peer example:

n 128 rules max, 15 translations per rule dial-peer voice 9003 voip
2. Create at least one voice translation profile and associated rules to incoming called-number 9003#.T
the profile: translation-profile incoming VOIP-out

voice-class codec 1
n 1000 profiles supported dtmf-relay h245-signal h245-alphanumeric
n Unique name for each profile

n You can apply one rule for each (calling numbers, called
numbers, and redirected called numbers).

[ 61 ]
COR CCVP GWGK Quick Reference Sheets by Brion S. Washington
COR Dial-peer cor list InternationalAccess

Member International
Class Of Restriction (COR) enables you to restrict incoming and outgo-
ing calls.
1. Define the COR label. dial-peer voice 1 pots
2. Build the permission groups. corlist outgoing LocalAccess
destination-pattern [2-9]..........
3. Apply the COR list to dial peers: port 1/0
a. Outbound dial peers, use corlist outgoing. !
b. Inbound dial peers, use corlist incoming. corlist outgoing LongAccess
destination-pattern 1...........
port 1/0
Define the COR Label !

Dial-peer cor custom
corlist outgoing InternationalAccess
Name local
destination-pattern 0.........
Name LongDistance
port 1/0
Name international
!
Build Permission Groups

Dial-peer cor list LocalAccess
Member Local
Dial-peer cor list LongAcess

Member LongDistance

[ 62 ]
ENTERPRISE VERSUS SER VICE PROVIDER CCVP GWGK Quick Reference Sheets by Brion S. Washington
Enterprise Versus Service

Provider
Two main types of VoIP networks are in use today: enterprise and
service provider. Each has similarities with the other, and each has
differences.
Enterprise networks are built by companies trying to cut costs and
leverage VoIP to help accomplish their business goals. Toll bypass is
one of the leading factors for these kinds of networks. The layout of the
networks can range from a single site to multiple sites, from centralized
control to distributed control. Enterprise networks usually use Internet
service provider circuits to complete their calls. They only go out on
the PSTN when in failover mode or after the call has reached a switch
local to the destination. Most of the equipment is owned by the
company, giving the company administrative authority over the devices
used.
Service provider networks are networks built to provide a service to

others for profit. Like enterprise networks, they can be a single site or
multiple sites. Service provider networks usually have a device located
locally to terminate the call. If they do not own the termination equip-
ment, they transparently route the call to another ITSP device. Billing
and accounting information are of utmost importance in provider
networks, because their main income is generated by their telephony
services. The equipment used are GWs, GKs, and softswitches. These
companies usually have no reason to implement voice messaging.


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CCVP GWGK Quick Reference Sheets Page 5 Return to Table of Contents

INTRODUCTION CCVP GWGK Quick Reference Sheets by Brion S. Washington

CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets

GATEKEEPERS CCVP GWGK Quick Reference Sheets by Brion S. Washington

Address translation: Translation of IP addresses to an E.164 phone

CCVP GWGK Quick Reference Sheets

GATEKEEPERS CCVP GWGK Quick Reference Sheets by Brion S. Washington

Bandwidth control: Provides endpoint bandwidth management Optional Functions

CCVP GWGK Quick Reference Sheets

FIGURE 1.1 Call Control and Setup Using a Gatekeeper

CCVP GWGK Quick Reference Sheets

AS diately. The GW sends a GRQ to the GK. A GK responds with a

gatekeeper confirm (GCF) or gatekeeper rejection (GRJ) message.

Multicast: The devices try to find a GK with the multicast IP

CCVP GWGK Quick Reference Sheets

Unregistration Process Types of Gatekeepers

CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets

You can use the following options with zone remote:

CCVP GWGK Quick Reference Sheets

Directory Gatekeepers Directory Gatekeeper Call Setup

Directory Gatekeeper Call Setup

V V The remaining steps are steps 4-10 in interzone scenario.

Directory Gatekeeper Configuration

The differences between implementing a directory GK instead of intra-

CCVP GWGK Quick Reference Sheets

Example: Directory Gatekeeper Configuration Gatekeeper Infantry Configuration

CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets

FIGURE 1.5 Interzone Topology Example, with Two Callers in Different

The difference between intrazone and interzone is the number of GKs

CCVP GWGK Quick Reference Sheets

Interzone Call Setup 1. User A dials 5577.

FIGURE 1.6 Interzone Call Steps

CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets

Gatekeeper Transaction GKTMP Messages

CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets

Gatekeeper Update Protocol n RegistrationIndication—Sent when a connection is made with a

GUP Messages Remote Zone Clustering

CCVP GWGK Quick Reference Sheets

Hot Standby Router Protocol ip address 75.0.0.1 255.0.0.0

CCVP GWGK Quick Reference Sheets

CCVP GWGK Quick Reference Sheets

VOICE GATEWAYS CCVP GWGK Quick Reference Sheets by Brion S. Washington

CCVP GWGK Quick Reference Sheets

VOICE GATEWAYS CCVP GWGK Quick Reference Sheets by Brion S. Washington

FIGURE 1.7 Simple Gateway Life

In the preceding diagram, the GW is not connected to another GW IP

FIGURE 1.8 GW-to-GW dating

This diagram shows a more complex GW network with a connection to

CCVP GWGK Quick Reference Sheets

VOICE GATEWAYS CCVP GWGK Quick Reference Sheets by Brion S. Washington

CCVP GWGK Quick Reference Sheets

VOICE GATEWAYS CCVP GWGK Quick Reference Sheets by Brion S. Washington

the Request URI or the locally configured proxy server. Setup

n Endpoint locating Connect ACK

n Dynamic endpoint registration Disconnect

n Toll bypass Release

FIGURE 1.10 SIP Call Flow