Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person real time traffic over the Web. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be expanded to deal with IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Kimball.
SIP is typically associated with a technique to provide the functionality of typical telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These older protocols functioned at a very low level to link IP phones to the public telephone system. SIP, however, provides a sophisticated and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the regular phone network.
SIP trunking provides multiple benefits to the business person such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the high subscription expenses of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by providing both voice and data over the very same connection. Businesses will have the manageability to route calls to favored carriers and the redundancy of using several service providers.
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