Cisco Systems H.323/SIP manual Setting Description Default, General Settings

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Chapter 3 Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Information About Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Note If you are using an IP PBX that runs standard H.323 or SIP call control, see the “How to Configure Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1” section on page 3-3for the required system settings and see your IP PBX documentation for information about how to configure those settings.

Table 3-1describes the Release 5.2.1 Management Console fields and lists the default settings.

Table 3-1 Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Management Console Fields and Default Settings

 

 

 

 

Setting

 

Description

 

Default

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

General Settings

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Max Number of

 

Maximum number of callers Release 5.2.1 will accept. This

 

960

 

 

 

 

 

Callers

 

maximum number can be a combination of H.323 and SIP

 

 

 

 

 

 

 

 

 

callers.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Outdial Protocol

Controls whether outdials from the IP-gateway server are

 

H.323

 

 

 

 

 

 

 

placed by using H.323 or SIP.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Note

In mixed H.323-SIP, call-control environments, you

 

 

 

 

 

 

 

 

 

 

 

must select one protocol for outdials; otherwise, the

 

 

 

 

 

 

 

 

 

 

 

default protocol will be used.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Verbose Logging

 

Sets the level of logging information.

 

Normal

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

H.323 Settings

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Enabled

 

Enables or disables the H.323 protocol.

 

Yes

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Max Number of

 

Maximum number of H.323 callers Release 5.2.1 accepts.

 

960

 

 

 

 

 

Callers

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

E.164 Address

 

A dialable number for the IP-gateway server.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

H323 ID

Caller ID name that is used by Release 5.2.1.

 

MeetingPlace

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Gateway Address and

 

IP address and port number of the server responsible for

 

Address: —

 

 

 

 

 

Gateway Port

 

routing H.323 calls. Outdials using H.323 are directed to this

 

Port: 1720

 

 

 

 

 

 

 

IP address and port if an H.323 gatekeeper is not used.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Note

You must enter this gateway information if you are

 

 

 

 

 

 

 

 

 

 

 

using H.323 without a gatekeeper.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Use Gatekeeper

Enables the IP-gateway server to register with an H.323

 

No

 

 

 

 

 

 

 

gatekeeper.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Gatekeeper Address

 

IP address and port number of the H.323 gatekeeper. If an

 

Address: —

 

 

 

 

 

and Gatekeeper Port

 

H.323 gatekeeper is used, Release 5.2.1 registers with the

 

Port: 1719

 

 

 

 

 

 

 

server and directs H.323 outdials to the server.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Note

If using an H.323 gatekeeper, ensure that your system

 

 

 

 

 

 

 

 

 

 

 

allows traffic to pass through ports 1024-65535

 

 

 

 

 

 

 

 

 

 

 

because MeetingPlace H.323/SIP IPGW uses these

 

 

 

 

 

 

 

 

 

 

 

ports for dynamic TCP and UDP traffic.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

3-2

 

 

 

 

 

 

 

 

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 Field Description Click the Add a New Gateway linkChoose 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 General Settings Setting Description DefaultSettings SIP Settings Field Name Setting SIP Field Name Setting Field Task Assigning the Primary IP AddressChoose 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 a Cisco IP Phone Dead Air Heard When Using an H.323 DeviceFast 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.