Cisco Systems H.323/SIP manual Description Value

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Appendix A Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Installation Worksheets

Information About the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial Plan

Description

 

Value

 

 

 

 

 

 

 

4. Additional IP addresses of the Cisco Unified

 

hostname

____________________________

 

 

 

MeetingPlace Audio Server system.

 

IP address

____________________________

 

 

 

 

 

 

 

 

 

Up to four additional IP addresses are needed

 

hostname

____________________________

 

 

 

for the Multi Access blade. If a TP1610 Multi

 

IP address

____________________________

 

 

 

Access blade is in use but only 240 VoIP or

 

 

 

 

fewer are deployed, then you must specify the

 

hostname

____________________________

 

 

 

lower address; the upper address can be set to

 

IP address

____________________________

 

 

 

0.0.0.0. You must also set the Ethernet switch

 

 

 

 

port or any other network devices to which the

 

hostname

____________________________

 

 

 

Multi Access blade connects directly to fixed

 

IP address

____________________________

 

 

 

100Base-TX Full Duplex.

 

 

 

 

 

 

 

 

 

 

 

Note

 

Do not set the lower address to 0.0.0.0.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

5. Hostname or IP address of one of the

 

hostname

____________________________

 

 

 

following:

 

IP address

____________________________

 

 

 

 

 

 

 

 

 

Cisco Unified CallManager server or

 

 

 

 

 

 

 

IP PBX that runs standard H.323 or SIP

 

 

 

 

 

 

 

call control

 

 

 

 

 

 

Cisco SIP Proxy Server

 

 

 

 

 

 

 

 

 

6. Host name or IP address of the Cisco Unified

 

hostname

____________________________

 

 

 

MeetingPlace Web Conferencing server if

 

IP address

____________________________

 

 

 

running on the same server as Release 5.2.1.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Note

 

If you use a hostname, DNS must be

 

 

 

 

 

 

 

enabled to resolve the hostname to an IP

 

 

 

 

 

 

 

address.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Information About the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial Plan

A dial plan ensures that IP and PSTN calls to and from the Cisco Unified MeetingPlace Audio Server system are directed to the proper endpoints on their respective network. Each type of call has a dial pattern that specifies its call flow to and from the MeetingPlace Audio Server system.

For example, if your Cisco Unified MeetingPlace Audio Server system has both IP and PSTN interfaces, you may want to configure their outdial patterns so that outdials to a PSTN phone will go through the Cisco Unified MeetingPlace Audio Server system PSTN interface. This ensures an outdial to a PSTN phone does not go through the IP network first and then to the PSTN.

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

 

A-2

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 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.