Cisco Systems H.323/SIP manual CallManager Administration

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Chapter 2 Installing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

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.

 

 

 

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

 

 

 

 

 

 

 

 

 

 

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OL-6571-02

 

 

 

 

 

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Contents Corporate Headquarters Text Part Number OL-6571-02Copyright 2005-2006 Cisco Systems, Inc. All rights reserved N T E N T S Fast Busy Signal Heard When Using a Cisco IP Phone Worksheets OL-6571-02 Audience ScopeNaming Conventions Used in This Guide New Features in This ReleaseFeature Description Product Naming ConventionCisco Unified MeetingPlace System SIP RTP Protocol Description SIPAudio Quality During a Cisco Unified MeetingPlace Meeting Step Cisco IP Phone Description Pstn Phone Description Step Cisco IP Phone Description Pstn Phone Description Step Device Description Cisco SIP IP Phone Description Additional References Step H.323 Device Description Cisco SIP IP Phone DescriptionA P T E R CallManager Administration Click the Add a New Gateway link Field DescriptionChoose Originator Field Description Task Server to the Cisco Unified CallManager Configuration Tone boxChoose Start Run Configuring Cisco Unified MeetingPlace Gateway SIM Select MeetingPlace IP Gateway and click RemoveChanging Cisco Unified MeetingPlace Gateway SIM Settings To apply the configuration settings, click OK again OL-6571-02 A P T E R Setting Description Default General SettingsSettings SIP Settings Field Name Setting SIP Field Name Setting Assigning the Primary IP Address Field TaskChoose Specify an IP address Information About Configuring a Dialing Group How to Configure a Dialing GroupConfiguring a Dialing Group Example OL-6571-02 Troubleshooting Network Connectivity Troubleshooting Caller Connectivity Unable to Make Calls From a Cisco IP PhoneDead Air Heard When Using an H.323 Device Dead Air Heard When Using a Cisco IP PhoneFast Busy Signal Heard When Using a Cisco IP Phone MP Resp. Msg=3 CPerr=0 SeqNum=0x16 Page Unable to Dial Out on IP Ports MeetingPlace IP outdial. Phone=651515 IRC=0 PSTN=46 Unit=0 Troubleshooting Audio Problems Poor or Low-Audio QualityEcho OL-6571-02 Worksheets Description ValueDescription Value MeetingPlace IP call flow Value OL-6571-02 D E IN-2 IN-3 IN-4

H.323/SIP specifications

Cisco Systems has been a leading force in the development and implementation of voice and video communication technologies, prominently featuring H.323 and SIP (Session Initiation Protocol). These protocols have become cornerstones in the realm of IP-based communication, facilitating seamless interaction across disparate devices and networks.

H.323 is a set of protocols that provides multimedia communication over packet-switched networks, such as the Internet. It supports audio, video, and data communications across IP networks, utilizing a variety of components including terminals, gateways, and multipoint control units (MCUs). One key feature of H.323 is its ability to handle both point-to-point and multipoint video conferencing. This makes it particularly suitable for enterprise applications where group communication is essential.

On the other hand, SIP is a more recent and flexible signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video, and messaging applications. Renowned for its simplicity and interoperability, SIP can work with various communication mediums and provides extensive capabilities for managing multimedia sessions. One of the primary advantages of SIP is its scalability, allowing it to accommodate everything from small-scale personal communications to vast corporate systems.

Both H.323 and SIP support features such as call transfer, call hold, and caller ID. H.323, however, can be more complex due to its broader array of standards and components, which might require substantial configuration. SIP, conversely, is designed to be lightweight, easily integrated with existing systems and applications, making it more user-friendly for developers.

Cisco Systems enhances these protocols through their robust telecommunication infrastructure that facilitates performance optimization, security, and unparalleled user experience. With advanced technologies like Cisco Unified Communications Manager and Cisco Webex, organizations can leverage H.323 and SIP to create cohesive communication environments. Security features such as encryption and authentication ensure that sensitive conversations remain private and secure, while Quality of Service (QoS) protocols manage bandwidth effectively to maintain consistent call quality.

Ultimately, the combination of H.323's established framework and SIP's flexibility ensures that organizations using Cisco Systems can effectively manage their communication needs, fostering collaboration and connectivity in today's fast-paced digital landscape. These protocols continue to evolve, adapting to the ever-increasing demands placed on modern communication systems.