Cisco Systems H.323/SIP manual Field Description Task

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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

Table 2-1 Fields in the Gateway Configuration Window (continued)

 

Field

Description

Task

 

Gatekeeper

Provides address translation and controls

Choose None.

 

Registration

access to the LAN for connections

 

 

 

 

between H.323-compliant devices, such as

 

 

 

 

terminals and gateways.

 

 

Media

Implements features that H.323 does not

Deselect this option.

 

Termination

support (such as hold and transfer) via

 

 

Point (MTP)

MTP. This check box is only for H.323

 

 

Required

clients and H.323 devices that do not

 

 

 

 

support the H.245 Empty Capabilities Set

 

 

 

 

message.

 

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.

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 ScopeNew Features in This Release Feature DescriptionNaming Conventions Used in This Guide 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 Choose Originator Click the Add a New Gateway linkField Description 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 Settings Setting Description DefaultGeneral Settings SIP Settings Field Name Setting SIP Field Name Setting Choose Specify an IP address Assigning the Primary IP AddressField Task 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 PhoneFast Busy Signal Heard When Using a Cisco IP Phone Dead Air Heard When Using an H.323 DeviceDead Air 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.
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