MeetingPlace IP call flow Value | Cisco Systems H.323/SIP guide

Appendix A Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Installation Worksheets

Information About the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial Plan

For Cisco Unified MeetingPlace Audio Servers systems that have both PSTN and IP interfaces, a dial plan should account for rollover from PSTN to IP ports and vice versa. For example, if you have a Cisco Unified MeetingPlace Audio Server system with 96 IP user licenses and 192 PSTN user licenses, the 97th caller to IP is automatically forwarded to a PSTN port by Cisco Unified CallManager through a voice gateway, rather than producing a fast busy signal.

For additional information about mixed-mode configuration, see the Configuration Guide for

Cisco Unified MeetingPlace Audio Server Release 5.3 at the following URL:

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

Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Dial Plan Worksheet

Use the following worksheet to create a a dial plan.

MeetingPlace IP call flow	Value

1. From an IP phone to the IP-gateway server.	dial pattern____________________
If the IP-gateway server is busy, Cisco Unified	A 4-digit number that does not conflict
CallManager can forward calls to Cisco Unified	with a corporate phone extension number
MeetingPlace system PSTN through a voice gateway.	scheme.
You must configure Cisco Unified CallManager and
the voice gateway to route this type of call.

2. From a PSTN phone to Cisco Unified MeetingPlace	dial pattern____________________
system PSTN.	A 7- or 10-digit phone number that does
	A 7- or 10-digit phone number that does
If Cisco Unified MeetingPlace system PSTN is busy,	not conflict with a corporate phone
the PBX or CO can forward calls to the IP-gateway	numbering scheme.
server through Cisco Unified CallManager. You must
configure the PBX or CO to route this type of call.

3. From Cisco Unified MeetingPlace system IP to an IP	dial pattern____________________
phone.	Typically, the last four digits of the phone
	Typically, the last four digits of the phone
	number.

4. From Cisco Unified MeetingPlace system PSTN to a	dial pattern____________________
PSTN phone.	Typically 9, if needed for an outside line,
	Typically 9, if needed for an outside line,
	followed by either the 7- or 10-digit phone
	number.

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

	OL-6571-02	A-3

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.