Cisco Systems H.323/SIP manual SIP Settings

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

How to Configure Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

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

 

 

 

 

 

 

 

SIP Settings

 

 

 

 

 

 

 

 

 

 

 

Enabled

 

Enables or disables the SIP protocol.

 

Yes

 

 

 

 

 

Max Number of

 

Maximum number of SIP callers Release 5.2.1 accepts.

 

960

Callers

 

 

 

 

 

 

 

 

 

 

Display Name

Display name of the IP-gateway server that is used for SIP

 

MeetingPlace

 

 

messages.

 

 

 

 

 

 

 

User Name

 

A dialable number for the IP-gateway server.

 

<blank>

 

 

 

 

Session Name

Session name used in Session Description Protocol (SDP)

 

MeetingPlace

 

 

body.

 

 

IP Call

 

 

 

 

 

Proxy Server Address

 

IP address and port number of the Cisco SIP Proxy Server.

 

Address: —

and Proxy Server Port

 

Cisco Unified MeetingPlace system outdials placed by using

 

Port: 5060

 

 

SIP are directed to this IP address and port.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Note

If using Cisco SIP Proxy Server, ensure that your

 

 

 

 

 

 

system allows traffic to pass through ports

 

 

 

 

 

 

1024-65535 because Release 5.2.1 uses these ports for

 

 

 

 

 

 

dynamic TCP and UDP traffic.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

How to Configure Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

You must configure Release 5.2.1 to dial out by using one of the following servers:

Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With Cisco Unified CallManager, page 3-4

Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With Cisco SIP Proxy Server, page 3-4

(Optional) Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 for Use With an H.323 Gatekeeper, page 3-5

(Optional) Verifying MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Configuration, page 3-6

Note Release 5.2.1 supports concurrent incoming H.323 and SIP calls; however, you must configure the Release 5.2.1 to use one protocol, either H.323 or SIP, to dial out.

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

 

OL-6571-02

3-3

 

 

 

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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 Choose Originator Click the Add a New Gateway linkField Description 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 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 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 ConnectivityFast 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 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.