Cisco Systems H.323/SIP manual Cisco Unified MeetingPlace System

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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 3-1for 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

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

 

OL-6571-02

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Contents Text Part Number OL-6571-02 Corporate HeadquartersCopyright 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 Scope AudienceFeature Description New Features in This ReleaseNaming Conventions Used in This Guide Product Naming ConventionCisco Unified MeetingPlace System SIP RTP SIP Protocol DescriptionAudio 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 Step H.323 Device Description Cisco SIP IP Phone Description Additional ReferencesA P T E R CallManager Administration Click the Add a New Gateway link Field DescriptionChoose Originator Field Description Task Tone box Server to the Cisco Unified CallManager ConfigurationChoose Start Run Select MeetingPlace IP Gateway and click Remove Configuring Cisco Unified MeetingPlace Gateway SIMChanging 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 How to Configure a Dialing Group Information About Configuring a Dialing GroupConfiguring a Dialing Group Example OL-6571-02 Troubleshooting Network Connectivity Unable to Make Calls From a Cisco IP Phone Troubleshooting Caller ConnectivityDead 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 Poor or Low-Audio Quality Troubleshooting Audio ProblemsEcho OL-6571-02 Description Value WorksheetsDescription 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.