Cisco Unified MeetingPlace H.323/SIP Service manual

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Administrator Guide
Cisco Unified MeetingPlace
H.323/SIP IP Gateway Software
Release 5.2.1
Revised: April 2006
Text Part Number: OL-6571-02

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Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Copyright © 2005-2006 Cisco Systems, Inc. All rights reserved.

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CONTENTS
CHAPTER 1Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 1-1
Audience 1-1
Scope 1-1
Naming Conventions Used in This Guide 1-2
New Features in This Release 1-2
Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Components 1-3
Cisco Unified MeetingPlace System 1-3
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 1-4
Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 1-4
Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Uses 1-5
Dual Tone Multi-Frequency Support by Cisco Unified MeetingPlace H.323/SIP IP Gateway
Software Release 5.2.1 1-5
Audio Quality During a Cisco Unified MeetingPlace Meeting 1-6
Endpoints That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 1-6
How PSTN and Cisco IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1 1-7
How H.323 Clients and Cisco SIP IP Phones Communicate by Using Cisco Unified MeetingPlace
H.323/SIP IP Gateway Software Release 5.2.1 1-9
Additional References 1-10
CHAPTER 2Installing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 2-1
How to Complete Prerequisites for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 Installation or Upgrade 2-1
How to Configure Cisco Unified CallManager for Use With Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1 2-2
Adding the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server to the
Cisco Unified CallManager Configuration Database 2-2
Assigning a Cisco Unified CallManager Route Pattern to Point to the Cisco Unified MeetingPlace
H.323/SIP IP Gateway Release Release 5.2.1 Server 2-4
How to Install or Upgrade to Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 2-5
Installing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 2-6

Contents
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OL-6571-02
Upgrading to Cisco Unified MeetingPlace H.323/SIP IPGW Software Release 5.2.1 From
Cisco Unified MeetingPlace IP Gateway Release 5.x2-6
Upgrading to Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 From
Cisco Unified MeetingPlace IP Gateway Release 4.x2-7
Configuring Cisco Unified MeetingPlace Gateway SIM 2-7
Changing Cisco Unified MeetingPlace Gateway SIM Settings 2-8
Uninstalling Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 2-9
CHAPTER 3Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 3-1
Information About Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release
5.2.1 3-1
How to Configure Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 3-3
Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With
Cisco Unified CallManager 3-4
Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With
Cisco SIP Proxy Server 3-4
Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With
an H.323 Gatekeeper 3-5
Verifying MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Configuration 3-6
Information About Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
for Use With Cisco Unified MeetingPlace Web Conferencing 3-7
How to Configure Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With
Cisco Unified MeetingPlace Web Conferencing 3-7
Assigning the Primary IP Address 3-7
Information About Configuring Multiple Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 Servers for Load Balancing and Redundancy 3-8
Information About Configuring a Dialing Group 3-8
How to Configure a Dialing Group 3-8
Configuring a Dialing Group Example 3-9
Information About Reservationless Single Number Access Configuration 3-9
Information About Reverse Connection to the MeetingPlace Audio Server System Configuration 3-10
CHAPTER 4Troubleshooting Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 4-1
Troubleshooting Network Connectivity 4-1
Troubleshooting Caller Connectivity 4-2
Unable to Make Calls From a Cisco IP Phone 4-2
Unable to Call a PSTN Telephone From a Cisco IP Phone or Vice Versa 4-2
Dead Air Heard When Using an H.323 Device 4-3
Dead Air Heard When Using a Cisco IP Phone 4-3
Fast Busy Signal Heard When Using a Cisco IP Phone 4-3

Contents
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Unable to Make Dial-Pad Key Selections When Using an H.323 Device 4-3
Checking the Cisco Unified MeetingPlace Audio Server System When IP Ports Do Not Answer 4-4
Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server
When IP Ports Do Not Answer 4-4
Checking Cisco Unified CallManager When IP Ports Do Not Answer 4-5
Checking the Cisco Unified MeetingPlace Audio Server System When IP Calls Connect But No Audio
Is Heard 4-5
Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 When IP
Calls Connect But No Audio Is Heard 4-6
Checking the Cisco IP Phone When IP Calls Connect But No Audio Is Heard 4-6
Unable to Dial Out on IP Ports 4-6
Checking the Cisco Unified MeetingPlace Audio Server System When Unable to Dial Out on IP
Ports 4-7
Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server
When Unable to Dial Out on IP Ports 4-7
Checking Cisco Unified CallManager When Unable to Dial Out on IP Ports 4-8
Troubleshooting Audio Problems 4-8
Poor or Low-Audio Quality 4-8
Echo 4-9
APPENDIX ACisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Installation
Worksheets A-1
Information About the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Installation Worksheet A-1
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Installation
Worksheet A-1
Information About the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial
Plan A-2
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial Plan
Worksheet A-3
I NDEX

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CHAPTER
1-1
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OL-6571-02
1
Introducing Cisco Unified MeetingPlace
H.323/SIP IP Gateway Software Release 5.2.1
This chapter includes the following sections:
•Audience, page 1-1
•Scope, page 1-1
•Naming Conventions Used in This Guide, page 1-2
•New Features in This Release, page 1-2
•Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Components, page 1-3
•Additional References, page 1-10
Note In this guide, Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 is referred to
as Release 5.2.1.
Audience
This guide is for network and telephony system administrators who are responsible for installing and
configuring Release 5.2.1 for use with the Cisco Unified MeetingPlace system.
Scope
Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
provides information about Release 5.2.1 that enables you to perform the following actions:
•Understand the Cisco Unified MeetingPlace system and related IP telephony components.
•Install and configure Release 5.2.1.
•Configure Cisco Unified CallManager to route IP calls to the IP-gateway server.
•Use Release 5.2.1 with IP PBX systems that are running standard H.323 or SIP call control—such
as Avaya, Nortel, Alcatel, and Pingtel systems.

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Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Naming Conventions Used in This Guide
This guide does not provide information about configuring third-party, call-control applications. If you
are using an IP PBX that runs standard H.323 or SIP call control, see the “Information About
Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1” section on
page 3-1 for required system settings and see your IP PBX documentation for information about how to
configure those settings.
Additionally, this guide does not provide information about installing Multi Access (MA) blades or
configuring the Cisco Unified MeetingPlace Audio Server system for IP; for more information about
these topics, see the “Additional References” section on page 1-10.
Naming Conventions Used in This Guide
The following naming conventions are used in this guide:
New Features in This Release
Release 5.2.1 includes the following new features:
Product Naming Convention
Cisco Unified MeetingPlace Audio Server release
and hardware upon which the release is installed
Cisco Unified MeetingPlace Audio Server system
Cisco Unified MeetingPlace Audio Server with
any possible combinations of integration
applications
Cisco Unified MeetingPlace system
Cisco Unified MeetingPlace Gateway System
Integrity Manager
Gateway SIM
Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1
Release 5.2.1
Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1—the hardware
upon which Release 5.2.1 is installed
IP-gateway server
Feature Description
Dialing Group Configuration Dialing group configuration customizes the Cisco Unified
MeetingPlace Audio Server system by presenting specific voice
prompts to callers who dial in to a meeting by using a particular IP
phone number.
Improved Cisco Unified
MeetingPlace Gateway SIM
Installation
During Release 5.2.1 installation, the Gateway SIM installs or
upgrades automatically if an earlier Gateway SIM release is detected.

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Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
Information About Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1 Components
Supporting up to 960 IP connections, Release 5.2.1 works with the Cisco Unified MeetingPlace Audio
Server system to provide meeting access to callers. The Cisco Unified MeetingPlace Audio Server
system supports connections from up to sixteen IP-gateway servers; this multigateway support provides
network load balancing and system redundancy.
To deploy Release 5.2.1, your network must have following system components:
•Cisco Unified MeetingPlace Audio Server system to provide conferencing functionality.
•Release 5.2.1 to perform IP call setup and tear down for the Cisco Unified MeetingPlace Audio
Server system.
•Endpoints that are supported by Release 5.2.1 to connect callers to the Cisco Unified MeetingPlace
Audio Server system.
•One of the following applications to route IP calls to the IP-gateway server:
– Cisco Unified CallManager
– Cisco SIP Proxy Server
– Cisco Gateway
Note If you are using an IP PBX that runs standard H.323 or SIP call control, see the “Information About
Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1” section on
page 3-1 for the required system settings and see your IP PBX documentation for information about how
to configure these settings.
Cisco Unified MeetingPlace System
Consisting of the Cisco Unified MeetingPlace Audio Server system and a variety of integration
applications, the Cisco Unified MeetingPlace system is an integrated communication and productivity
tool that is deployed on a corporate network behind the firewall. With the Cisco Unified MeetingPlace
system, users in different locations can collaborate in real time by sharing documents over personal
computers and discussing content over telephones.
Access to the Cisco Unified MeetingPlace system is easy through end-user desktop applications, such
as web browsers and instant messaging clients. The Cisco Unified MeetingPlace system also integrates
with groupware clients and PSTN and IP-based telephones. Because of this access and integration, users
can quickly schedule and attend Cisco Unified MeetingPlace meetings from any location by using their
preferred interfaces.
For additional information about the Cisco Unified MeetingPlace system, see the Installation Planning
Guide for Cisco Unified MeetingPlace 5.3 at the following URL:
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_installation_guides_list.html

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Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
IP telephony uses your data network infrastructure to transmit voice packets. The underlying technology
that is used by IP telephony applications is Voice over IP (VoIP), which enables different types of
endpoints—IP phones, PSTN phones, and H.323 clients, for example—to communicate over your
network.
The following sections provide information about VoIP concepts and how they relate to Release 5.2.1:
•Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1, page 1-4
•Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Uses,
page 1-5
•Dual Tone Multi-Frequency Support by Cisco Unified MeetingPlace H.323/SIP IP Gateway
Software Release 5.2.1, page 1-5
•Audio Quality During a Cisco Unified MeetingPlace Meeting, page 1-6
Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1
Release 5.2.1 supports the following networking and telephony standards:
•H.323
•SIP
•RTP
•Codec G.711 alaw and ulaw (64 kbps) and G.729a (8 kbps)
Note By default, G.729a is not enabled, and G711 codec calls are negotiated first. For more
information about assigning codec preferences, see the Configuration Guide for Cisco Unified
MeetingPlace Audio Server Release 5.3 at the following URL:
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_installation_and_configura
tion_guides_list.html

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Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Uses
Protocols are rules that endpoints follow for sending and receiving messages, checking errors, and
compressing data. Release 5.2.1 uses the following protocols to transmit data throughout the
Cisco Unified MeetingPlace system:
Dual Tone Multi-Frequency Support by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1
Dual Tone Multi-Frequency (DTMF) is a signaling method that allocates a specific pair of frequencies
to each key on a touch-tone telephone. Various Cisco Unified MeetingPlace Audio Server system
functions are invoked when callers press touch-tone keys in certain combinations. For example, the #5
key combination enables callers to mute and unmute their phones during a meeting.
PSTN phones use in-band DTMF, which embeds the tone in the audio stream. Although in-band DTMF
is efficient, it cannot carry DTMF signals reliably when a voice compression codec is used.
Protocol Description
H.323 The protocol that is responsible for communication between
Cisco Unified CallManager and Release 5.2.1. The protocol suite,
which extends H.225 for call signaling and H.245 for data transfer, is
used in the successful acceptance and media exchange of data.
Session Initiation Protocol
(SIP)
A call-control protocol that supports all existing functionality that is
available to a Cisco IP phone. Release 5.2.1 complies with RFC 3261
and RFC 3515 specifications and interoperates with the following
endpoints:
•Cisco SIP Proxy Server environment
•Cisco 7960 and Cisco 7940 SIP IP phones
•Cisco IP/Videoconferencing Multipoint Control Unit
(IP/VC MCU)
•Microsoft Real-Time Communications (RTC) Server for
integration with Windows XP Messenger
Real-Time Transport
Protocol (RTP)
An Internet protocol responsible for the transmission of real-time data,
such as video and audio. Generally, RTP runs on top of User Datagram
Protocol (UDP) but can also be supported by other transport protocols.
For Release 5.2.1, RTP is responsible for carrying the G.711 and
G.729a encoded data. G.711 is a standard 64 kbps codec, and G.729a is
an 8 kbps codec. Both codecs offer quality audio transmission over
high-speed connections.
Skinny Station Protocol
(SSP)
A protocol that is used to establish connections, locate resources,
forward data, and handle flow control and error recovery, which enable
a Cisco IP phone to notify Cisco Unified CallManager of its ability to
place and receive calls.
Cisco Unified MeetingPlace
Gateway System Integrity
Manager (SIM)
A messaging service that enables NT services on the IP-gateway server
to communicate directly with the Cisco Unified MeetingPlace system.

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Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
H.323 clients can use out-of-band DTMF, which carries digitized information on a separate data channel
and sends this information directly to Release 5.2.1. Because out-of-band DTMF does not require that
the tone be deciphered, distortion and signal loss are minimal.
The Cisco Unified MeetingPlace system also supports RFC 2833: DTMF signals can be sent in the RTP
stream by using packets designed to carry the signal characteristics. The DTMF signal is not embedded
in the media and, therefore, does not suffer signal loss due to audio compression.
Release 5.2.1 handles both in-band and out-of-band DTMF.
Note Release 5.2.1 does not support out-of-band digit detection with SIP.
Audio Quality During a Cisco Unified MeetingPlace Meeting
The audio quality during a meeting depends upon the architecture of your network. Severe demands on
bandwidth, overloading, and latency cause dropped packets, resulting in broken audio, congestion, and
disruption of service.
In general, a switched-100 Mbps network handles VoIP traffic efficiently. To alleviate potentially
disruptive service and to improve audio quality, consider implementing class of service (CoS) and
quality of service (QoS).
When the server handles over 400 ports of IP calls, voice quality degradation can occur because of
network congestion. CoS is a technology that helps manage network traffic by assigning a class to
similar types of traffic and assigning a priority to each class. Typically in a VoIP environment, voice
traffic is set to a high priority while data traffic is set to a low priority, and CoS makes a best effort to
provide QoS by managing traffic based upon the assigned class and priority.
Release 5.2.1 implements IP Precedence Level 5 CoS for voice traffic. If your network is set to use this
CoS, the resulting QoS maximizes audio quality during your meetings.
Note Release 5.2.1 does not support sending Layer 2 QoS or CoS; therefore, you cannot set priorities at the
Layer 2 switch level.
Endpoints That are Supported by Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1
Release 5.2.1 integrates easily with existing networks to host Cisco Unified MeetingPlace meetings for
users through the following supported endpoints:
•Cisco IP Phones
•Cisco SIP IP Phones
•H.323 clients, such as Microsoft NetMeeting
•PSTN phones through a voice gateway

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Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
How PSTN and Cisco IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1
When a call is placed from a PSTN phone to a Cisco IP phone, the call is routed through a voice gateway,
which is the demarcation point where the circuit-switched voice network meets the packet-switched data
network. The primary responsibility of the voice gateway is to ensure that PSTN voice traffic reaches
the data network and vice versa. You can use the voice gateway to forward an IP or PSTN call to its
opposing network through Cisco Unified CallManager or a PBX.
When a call is placed from an Cisco IP phone, it is routed to Cisco Unified CallManager, which is
responsible for setting up the call, directing the call to the called device, and sending network
information— such as the IP address, UDP port number, and communication capabilities of the called
device—to the Cisco IP phone. After receiving the information, the Cisco IP phone sends its digitized
voice traffic directly to the called device.
The following steps describe how Cisco IP phones and PSTN phones use Release 5.2.1 to access the
Cisco Unified MeetingPlace Audio Server system, as shown in Figure 1-1.
Figure 1-1 Cisco IP Phones and PSTN Phones Using Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software to Access the Cisco Unified MeetingPlace Audio Server System
.
Step Cisco IP Phone Description PSTN Phone Description
1. On the Cisco IP phone dial pad, the caller enters a
dialable number to the Cisco Unified
MeetingPlace Audio Server system that will host
the meeting.
By using a PSTN phone, the caller dials the number
to the voice gateway.
2. The call is immediately routed by using SSP to
Cisco Unified CallManager.
The voice gateway routes the call to Cisco Unified
CallManager.
Cisco IP phone
Cisco CallManager
PSTN phone
Cisco
MeetingPlace
Audio Server
Cisco MeetingPlace
H.323/SIP IP Gateway
Voice gateway
121557
1
2
3
4
2
3
4
1
5
5
5
4
3
4
3
V
IP
IP

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3. Cisco Unified CallManager and Release 5.2.1
communicate by using H.323. This
communication process involves H.225 for call
signaling and H.245 for media exchange.
Cisco Unified CallManager examines its routing
table to resolve the dialed number with the IP
address of the IP-gateway server.
Cisco Unified CallManager and Release 5.2.1
communicate by using H.323. This communication
process involves H.225 for call signaling and H.245
for media exchange.
a. Cisco Unified CallManager and Release 5.2.1 use H.225 to determine if the Cisco Unified
MeetingPlace Audio Server system can accept the call. By using Cisco Unified MeetingPlace
GWSIM, Release 5.2.1 communicates directly with the Cisco Unified MeetingPlace Audio
Server system to determine its availability.
b. If the Cisco Unified MeetingPlace Audio Server system is unavailable, Release 5.2.1 informs
Cisco Unified CallManager, and the caller hears a fast busy signal.
c. If the call is accepted, Cisco Unified CallManager and Release 5.2.1 use H.245 to negotiate
which codec will carry the voice activity. Release 5.2.1 uses G.711 or G.729a to carry the
encoded speech.
d. Once codec negotiation is complete, Release 5.2.1 uses the Gateway SIM to retrieve an IP
address and UDP port number from the Cisco Unified MeetingPlace Audio Server system. This
IP address and UDP port number provide access to the meeting.
4. Cisco Unified CallManager and Release 5.2.1 exchange the IP address and UDP port number of the
Cisco IP phone or voice gateway and the Cisco Unified MeetingPlace Audio Server system
a. Cisco Unified CallManager sends the IP address and UDP port number of the Cisco Unified
MeetingPlace Audio Server system to the Cisco IP phone or voice gateway.
b. Release 5.2.1 sends the IP address and UDP port number of the Cisco IP phone or voice gateway
to the Cisco Unified MeetingPlace Audio Server system.
5. After codec information, IP address, and UDP port number are received, the Cisco IP phone or voice
gateway uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio
Server system. The Cisco IP phone or voice gateway is connected to the Cisco Unified MeetingPlace
Audio Server system after each device exchanges data.
Step Cisco IP Phone Description PSTN Phone Description

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Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
How H.323 Clients and Cisco SIP IP Phones Communicate by Using Cisco Unified MeetingPlace
H.323/SIP IP Gateway Software Release 5.2.1
H.323 clients and Cisco SIP IP phones—which can be simultaneously deployed—communicate with
Release 5.2.1 and provide another option to join a Cisco Unified MeetingPlace meeting.
The following steps describe how H.323 devices and Cisco SIP IP phones access the Cisco Unified
MeetingPlace Audio Server system by using Release 5.2.1.
Figure 1-2 H.323 Device and Cisco SIP IP Phone Using Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software to Access the Cisco Unified MeetingPlace Audio Server System
.
Step H.323 Device Description Cisco SIP IP Phone Description
1. A caller places a call from an H.323 device
interface.
A caller places a call from a Cisco SIP IP phone.
2. The H.323 device and Release 5.2.1 communicate
by using H.323.
The Cisco SIP IP phone through Cisco SIP Proxy
Server and Release 5.2.1 communicate by using SIP.
a. The H.323 device or Cisco SIP IP phone and Release 5.2.1 determine if the Cisco Unified
MeetingPlace Audio Server system can accept the call. By using the Gateway SIM, the
Release 5.2.1 communicates directly with the Cisco Unified MeetingPlace Audio Server system
to determine its availability.
b. If the Cisco Unified MeetingPlace Audio Server system is unavailable, Release 5.2.1 informs
the H.323 device or Cisco SIP IP phone, and depending upon system configuration, callers may
hear a message informing them that the call cannot be accepted.
c. If the call is accepted, the H.323 device or Cisco SIP IP phone and Release 5.2.1 negotiate
which codec will carry the voice activity. Release 5.2.1 uses G.711 or G.729a to carry the
encoded speech.
d. Once codec negotiation is complete, Release 5.2.1 retrieves an IP address and UDP port number
from the Cisco Unified MeetingPlace Audio Server system by using Gateway SIM. This IP
address and UDP port number provide access to the meeting.
IP IP
H.323 device
Cisco SIP
IP phone
Cisco MeetingPlace
H.323/SIP IP Gateway
Cisco
MeetingPlace
Audio Server
Cisco SIP
proxy server
121556
1
2
2
1
4
4
4
3
2

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Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Additional References
Additional References
See to the following documents for additional information:
•Administrator Guide for Cisco Unified MeetingPlace Audio Server Release 5.3
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_maintenance_guides_list.html
•Cisco Unified CallManager documentation for your release
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/index.htm
•Cisco SIP Proxy Server documentation for your release
http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/index.htm
•Configuration Guide for Cisco Unified MeetingPlace Audio Server Release 5.3
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_installation_and_configuration
_guides_list.html
•Guide to Cisco Unified MeetingPlace Conferencing Documentation and Support
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_documentation_roadmaps_list.
html
•Installation Planning Guide for Cisco Unified MeetingPlace Release 5.3
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_installation_guides_list.html
•Release Notes for Cisco Unified MeetingPlace Audio Server Release 5.3
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_release_notes_list.html
•Release Notes for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_release_notes_list.html
3. The H.323 device or Cisco SIP IP phone and Release 5.2.1 exchange IP addresses and UDP port
numbers.
a. Release 5.2.1 sends the IP address and UDP port number of the Cisco Unified MeetingPlace
Audio Server system to the H.323 device or Cisco SIP IP phone.
b. Release 5.2.1 sends the IP address and UDP port number of the H.323 device or Cisco SIP IP
phone to the Cisco Unified MeetingPlace Audio Server system.
4. After codec information, IP address, and UDP port number of the Cisco Unified MeetingPlace Audio
Server system are received, the H.323 device or Cisco SIP IP phone uses the information to send voice
traffic directly to the Cisco Unified MeetingPlace Audio Server system. The H.323 device or
Cisco SIP IP phone is connected to the Cisco Unified MeetingPlace Audio Server system after each
device exchanges data.
Step H.323 Device Description Cisco SIP IP Phone Description

CHAPTER
2-1
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2
Installing Cisco Unified MeetingPlace H.323/SIP
IP Gateway Software Release 5.2.1
To install Release 5.2.1, perform the following procedures in this order:
•How to Complete Prerequisites for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 Installation or Upgrade, page 2-1
•How to Configure Cisco Unified CallManager for Use With Cisco Unified MeetingPlace H.323/SIP
IP Gateway Software Release 5.2.1, page 2-2
•How to Install or Upgrade to Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1, page 2-5
How to Complete Prerequisites for Cisco Unified MeetingPlace
H.323/SIP IP Gateway Software Release 5.2.1 Installation or
Upgrade
•Verify that your system meets the requirements listed in the Release Notes for Cisco Unified
MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1.
•Complete the “Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Installation Worksheet” section on page A-1 and “Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software Release 5.2.1 Dial Plan Worksheet” section on page A-3.
These worksheets identify the required information that you need to install and configure Release
5.2.1 to work with VoIP devices.
•By following the instructions in the “How to Configure Cisco Unified CallManager for Use With
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1” section on page 2-2,
configure Cisco Unified CallManager for your network.

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How to Configure Cisco Unified CallManager for Use With Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
•If a firewall separates the Cisco Unified MeetingPlace Audio Server system from the IP-gateway
server, open port 5003.
Tip The Gateway SIM communicates with the Cisco Unified MeetingPlace Audio Server system through
port 5003. This port can be bidirectional or unidirectional and can be opened on either the Cisco Unified
MeetingPlace Audio Server system or the IP-gateway server depending on your corporate security
needs.
•Stop all previously installed Cisco Unified MeetingPlace system services.
How to Configure Cisco Unified CallManager for Use With
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1
When a caller dials a number from an IP phone, the call is first directed to Cisco Unified CallManager;
from there, Cisco Unified CallManager associates the dialed number with a route pattern that points to
the appropriate IP-gateway server.
Note Traffic must be allowed to pass through ports 1024-65535 because the IP-gateway server uses these
ports to send dynamic TCP and UDP traffic to Cisco Unified CallManager.
Before you can install and configure Release 5.2.1, you must configure Cisco Unified CallManager to
point to your IP-gateway server. To configure Cisco Unified CallManager, you must first add a gateway;
the, assign the gateway to a route pattern.
To configure Cisco Unified CallManager for use with Release 5.2.1, perform the following procedures
in this order:
•Adding the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server to the
Cisco Unified CallManager Configuration Database, page 2-2
•Assigning a Cisco Unified CallManager Route Pattern to Point to the Cisco Unified MeetingPlace
H.323/SIP IP Gateway Release Release 5.2.1 Server, page 2-4
Adding the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1 Server to the Cisco Unified CallManager Configuration Database
To enable Cisco Unified CallManager to route calls to IP-gateway servers in your network, you must first
add each IP-gateway server to the Cisco Unified CallManager configuration database.
Step 1 From the Cisco Unified CallManager server, choose Start > Programs > Cisco Unified CallManager
> CallManager Administration.
Step 2 Enter the user name and password in the appropriate fields and click OK.
Step 3 In the Cisco Unified CallManager Administration page, choose System > CallManager.
Step 4 To display the Find/List Gateways window, choose Device > Gateway.

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Step 5 Click the Add a New Gateway link.
The Add a New Gateway window appears.
Step 6 From the Gateway drop-down menu, choose H.323 Gateway.
In the Device Protocol drop-down menu, the H.225 device protocol appears.
Step 7 Click Next.
The Gateway Configuration window appears.
C.
Step 8 Enter information in each field of the Gateway Configuration window, as shown in Table 2-1.
Note Maintain the default setting for all other parameters
.
Table 2-1 Fields in the Gateway Configuration Window
Field Description Task
Device Name Identifies the Cisco Unified CallManager
device.
Enter the hostname or IP address of the
IP-gateway server.
Device Pool Specifies a collection of properties for this
device including Cisco Unified
CallManager Group, Date/Time Group,
Region, and Calling Search Space for
autoregistration of devices.
Choose Default.
Locations Specifies the total bandwidth that is
available for calls to and from this
location. A location setting of None means
that the locations feature does not keep
track of the bandwidth that is consumed by
this device.
If applicable, choose the location of the
IP-gateway server on your network.
Calling Party
Selection
Sends directory number information for an
outbound call. Information in this field
determines which directory number is
sent. The following options specify which
directory number is sent:
•Originator—Sends the directory
number of the calling device.
•First Redirect Number—Sends the
directory number of the redirecting
device.
•Last Redirect Number—Sends the
directory number of the last device to
redirect the call.
Choose Originator.
Presentation
Bit
Determines whether the central office
transmits or blocks caller ID.
Choose None.

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How to Configure Cisco Unified CallManager for Use With Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Step 9 Click Insert.
Assigning a Cisco Unified CallManager Route Pattern to Point to the
Cisco Unified MeetingPlace H.323/SIP IP Gateway Release Release 5.2.1
Server
After adding the IP-gateway server to the Cisco Unified CallManager configuration database, you must
assign a route pattern, which comprises a string of digits (an address) and a set of associated digit
manipulations that can be assigned to the IP-gateway server. Route patterns work with route filters and
route lists to direct calls to the IP-gateway server and to include, exclude, or modify specific digit
patterns.
Tip Assigning 8XXX to a gateway routes all directory numbers 8000 to 8999 out the gateway. Similarly,
82XX routes directory numbers 8200 to 8299.
Step 1 If applicable, ensure that you have configured the following items in Cisco Unified CallManager:
•Gateway
•Route list
•Partition
•Route filter
Step 2 In the Cisco Unified CallManager Administration page, choose Route Plan > Route Pattern.
Step 3 Click Add a New Route Pattern.
Gatekeeper
Registration
Provides address translation and controls
access to the LAN for connections
between H.323-compliant devices, such as
terminals and gateways.
Choose None.
Media
Termination
Point (MTP)
Required
Implements features that H.323 does not
support (such as hold and transfer) via
MTP. This check box is only for H.323
clients and H.323 devices that do not
support the H.245 Empty Capabilities Set
message.
Deselect this option.
Table 2-1 Fields in the Gateway Configuration Window (continued)
Field Description Task