Cisco Systems H.323/SIP manual Additional References

Page 16

Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Additional References

Step H.323 Device Description

Cisco SIP IP Phone Description

 

 

3.The H.323 device or Cisco SIP IP phone and Release 5.2.1 exchange IP addresses and UDP port numbers.

a.Release 5.2.1 sends the IP address and UDP port number of the Cisco Unified MeetingPlace Audio Server system to the H.323 device or Cisco SIP IP phone.

b.Release 5.2.1 sends the IP address and UDP port number of the H.323 device or Cisco SIP IP phone to the Cisco Unified MeetingPlace Audio Server system.

4.After codec information, IP address, and UDP port number of the Cisco Unified MeetingPlace Audio Server system are received, the H.323 device or Cisco SIP IP phone uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio Server system. The H.323 device or

Cisco SIP IP phone is connected to the Cisco Unified MeetingPlace Audio Server system after each device exchanges data.

Additional References

See to the following documents for additional information:

Administrator Guide for Cisco Unified MeetingPlace Audio Server Release 5.3 http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_maintenance_guides_list.html

Cisco Unified CallManager documentation for your release http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/index.htm

Cisco SIP Proxy Server documentation for your release http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/index.htm

Configuration Guide for Cisco Unified MeetingPlace Audio Server Release 5.3

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_installation_and_configuration _guides_list.html

Guide to Cisco Unified MeetingPlace Conferencing Documentation and Support

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_documentation_roadmaps_list. html

Installation Planning Guide for Cisco Unified MeetingPlace Release 5.3 http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_installation_guides_list.html

Release Notes for Cisco Unified MeetingPlace Audio Server Release 5.3 http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_release_notes_list.html

Release Notes for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_release_notes_list.html

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

1-10

OL-6571-02

 

 

Image 16
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.