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

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

When a call is placed from a PSTN phone to a Cisco IP phone, the call is routed through a voice gateway, which is the demarcation point where the circuit-switched voice network meets the packet-switched data network. The primary responsibility of the voice gateway is to ensure that PSTN voice traffic reaches the data network and vice versa. You can use the voice gateway to forward an IP or PSTN call to its opposing network through Cisco Unified CallManager or a PBX.

When a call is placed from an Cisco IP phone, it is routed to Cisco Unified CallManager, which is responsible for setting up the call, directing the call to the called device, and sending network information— such as the IP address, UDP port number, and communication capabilities of the called device—to the Cisco IP phone. After receiving the information, the Cisco IP phone sends its digitized voice traffic directly to the called device.

The following steps describe how Cisco IP phones and PSTN phones use Release 5.2.1 to access the Cisco Unified MeetingPlace Audio Server system, as shown in Figure 1-1.

Figure 1-1 Cisco IP Phones and PSTN Phones Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software to Access the Cisco Unified MeetingPlace Audio Server System

Cisco IP phone

 

IP

 

3

1

5

 

4

2

Cisco MeetingPlace

H.323/SIP IP Gateway

33

 

Cisco CallManager

4

4

 

 

 

 

 

2

4

 

 

 

3

 

1

 

5

 

 

 

IP

V

 

 

 

 

 

PSTN phone

5

 

 

Voice gateway

 

 

 

Cisco

MeetingPlace

Audio Server

121557

.

Step

Cisco IP Phone Description

PSTN Phone Description

 

 

 

1.

On the Cisco IP phone dial pad, the caller enters a

By using a PSTN phone, the caller dials the number

 

dialable number to the Cisco Unified

to the voice gateway.

 

MeetingPlace Audio Server system that will host

 

 

the meeting.

 

 

 

 

2.

The call is immediately routed by using SSP to

The voice gateway routes the call to Cisco Unified

 

Cisco Unified CallManager.

CallManager.

 

 

 

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 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 Field Description Click the Add a New Gateway linkChoose 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 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 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 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 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.
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