Cisco Systems H.323/SIP manual Step Cisco IP Phone Description Pstn Phone Description

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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

Step

Cisco IP Phone Description

PSTN Phone Description

 

 

 

3.

Cisco Unified CallManager and Release 5.2.1

Cisco Unified CallManager examines its routing

 

communicate by using H.323. This

table to resolve the dialed number with the IP

 

communication process involves H.225 for call

address of the IP-gateway server.

 

signaling and H.245 for media exchange.

Cisco Unified CallManager and Release 5.2.1

 

 

 

 

communicate by using H.323. This communication

 

 

process involves H.225 for call signaling and H.245

 

 

for media exchange.

 

 

 

a.Cisco Unified CallManager and Release 5.2.1 use H.225 to determine if the Cisco Unified MeetingPlace Audio Server system can accept the call. By using Cisco Unified MeetingPlace GWSIM, Release 5.2.1 communicates directly with the Cisco Unified MeetingPlace Audio Server system to determine its availability.

b.If the Cisco Unified MeetingPlace Audio Server system is unavailable, Release 5.2.1 informs Cisco Unified CallManager, and the caller hears a fast busy signal.

c.If the call is accepted, Cisco Unified CallManager and Release 5.2.1 use H.245 to negotiate which codec will carry the voice activity. Release 5.2.1 uses G.711 or G.729a to carry the encoded speech.

d.Once codec negotiation is complete, Release 5.2.1 uses the Gateway SIM to retrieve an IP address and UDP port number from the Cisco Unified MeetingPlace Audio Server system. This IP address and UDP port number provide access to the meeting.

4.Cisco Unified CallManager and Release 5.2.1 exchange the IP address and UDP port number of the Cisco IP phone or voice gateway and the Cisco Unified MeetingPlace Audio Server system

a.Cisco Unified CallManager sends the IP address and UDP port number of the Cisco Unified MeetingPlace Audio Server system to the Cisco IP phone or voice gateway.

b.Release 5.2.1 sends the IP address and UDP port number of the Cisco IP phone or voice gateway to the Cisco Unified MeetingPlace Audio Server system.

5.After codec information, IP address, and UDP port number are received, the Cisco IP phone or voice gateway uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio Server system. The Cisco IP phone or voice gateway is connected to the Cisco Unified MeetingPlace Audio Server system after each device exchanges data.

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

1-8

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 ScopeNaming Conventions Used in This Guide New Features in This ReleaseFeature Description 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.