Cisco Systems H323-SIP User Manual

Administrator Guide
Cisco Unified MeetingPlace
H.323/SIP IP Gateway Software

Release 5.2.1
Revised: April 2006

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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.

CONTENTS

CHAPTER

1 Introducing 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

CHAPTER

OL-6571-02

Additional References 1-10

2 Installing 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

Administrator Guide for Cisco Unified MeetingPlace H.323 SIP/IP Gateway Software Release 5.2.1

iii

Contents

Upgrading to Cisco Unified MeetingPlace H.323/SIP IPGW Software Release 5.2.1 From
Cisco Unified MeetingPlace IP Gateway Release 5.x

2-6

Upgrading to Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 From
Cisco Unified MeetingPlace IP Gateway Release 4.x

2-7

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

3 Configuring 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

CHAPTER

iv

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

4 Troubleshooting 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

Administrator Guide for Cisco Unified MeetingPlace H.323 SIP/IP Gateway Software Release 5.2.1

OL-6571-02

Contents

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

APPENDIX

I

NDEX

Troubleshooting Audio Problems 4-8

Poor or Low-Audio Quality 4-8
Echo 4-9

A Cisco 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

A-2

Plan

Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial Plan
Worksheet

A-3

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Administrator Guide for Cisco Unified MeetingPlace H.323 SIP/IP Gateway Software Release 5.2.1

v

Contents

vi

Administrator Guide for Cisco Unified MeetingPlace H.323 SIP/IP Gateway Software Release 5.2.1

OL-6571-02

CHA P TER

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

Scope

OL-6571-02

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.

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.

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

1-1

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:

Product Naming Convention

Cisco Unified MeetingPlace Audio Server release
and hardware upon which the release is installed

Cisco Unified MeetingPlace Audio Server with
any possible combinations of integration
applications

Cisco Unified MeetingPlace Gateway System
Integrity Manager

Cisco Unified MeetingPlace H.323/SIP IP
Gateway Software 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

Cisco Unified MeetingPlace Audio Server system

Cisco Unified MeetingPlace system

Gateway SIM

Release 5.2.1

IP-gateway server

New Features in This Release

Release 5.2.1 includes the following new features:

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

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

1-2

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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1-3

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

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

1-4

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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:

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)

Real-Time Transport
Protocol (RTP)

Skinny Station Protocol
(SSP)

Cisco Unified MeetingPlace
Gateway System Integrity
Manager (SIM)

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

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.

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.

A messaging service that enables NT services on the IP-gateway server
to communicate directly with 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.

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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

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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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

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

Cisco IP phone

IP

5

1

3

2

4

Cisco CallManager

1

IP

PSTN phone

.

5

Voice gateway

Cisco MeetingPlace

H.323/SIP IP Gateway

3

4

2

V

4

3

3

4

5

Cisco
MeetingPlace
Audio Server

121557

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

By using a PSTN phone, the caller dials the number

to the voice gateway.
MeetingPlace Audio Server system that will host
the meeting.

2. The call is immediately routed by using SSP to

Cisco Unified CallManager.

The voice gateway routes the call to Cisco Unified

CallManager.

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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

Step Cisco IP Phone Description PSTN Phone Description

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.

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.

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.

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.

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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

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

H.323 device

Cisco MeetingPlace

H.323/SIP IP Gateway

1

IP

Cisco SIP

IP phone

.

4

IP

Cisco SIP

proxy server

1

2

2

4

2

3

4

Cisco
MeetingPlace
Audio Server

121556

Step H.323 Device Description Cisco SIP IP Phone Description

1. A caller places a call from an H.323 device

A caller places a call from a Cisco SIP IP phone.
interface.

2. The H.323 device and Release 5.2.1 communicate

by using H.323.

a. The H.323 device or Cisco SIP IP phone and Release 5.2.1 determine if the Cisco Unified

The Cisco SIP IP phone through Cisco SIP Proxy

Server and Release 5.2.1 communicate by using SIP.

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.

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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.

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

1-9

Additional References

Step H.323 Device Description Cisco SIP IP Phone Description

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.

Additional References

See to the following documents for additional information:

Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

• 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

1-10

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

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CHA P TER

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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Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

2-1