Cisco Systems H.323/SIP manual Step Device Description Cisco SIP IP 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

How H.323 Clients and Cisco SIP IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

H.323 clients and Cisco SIP IP phones—which can be simultaneously deployed—communicate with Release 5.2.1 and provide another option to join a Cisco Unified MeetingPlace meeting.

The following steps describe how H.323 devices and Cisco SIP IP phones access the Cisco Unified MeetingPlace Audio Server system by using Release 5.2.1.

Figure 1-2

H.323 Device and Cisco SIP IP Phone Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software to Access the Cisco Unified MeetingPlace Audio Server System

IP

Cisco SIP IP phone

H.323 device

1

2

Cisco MeetingPlace H.323/SIP IP Gateway

2

1

IP 4

Cisco SIP

proxy server

4

2

Cisco

 

MeetingPlace

3

Audio Server

 

4

121556

.

Step

H.323 Device Description

Cisco SIP IP Phone Description

 

 

 

1.

A caller places a call from an H.323 device

A caller places a call from a Cisco SIP IP phone.

 

interface.

 

 

 

 

2.

The H.323 device and Release 5.2.1 communicate

The Cisco SIP IP phone through Cisco SIP Proxy

 

by using H.323.

Server and Release 5.2.1 communicate by using SIP.

 

 

 

a.The H.323 device or Cisco SIP IP phone and Release 5.2.1 determine if the Cisco Unified MeetingPlace Audio Server system can accept the call. By using the Gateway SIM, the 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 the H.323 device or Cisco SIP IP phone, and depending upon system configuration, callers may hear a message informing them that the call cannot be accepted.

c.If the call is accepted, the H.323 device or Cisco SIP IP phone and Release 5.2.1 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 retrieves an IP address and UDP port number from the Cisco Unified MeetingPlace Audio Server system by using Gateway SIM. This IP address and UDP port number provide access to the meeting.

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 AudienceProduct Naming Convention New Features in This ReleaseFeature Description Naming Conventions Used in This GuideCisco 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.