Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transmission technology for person-to-person real time data over the Web. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides 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 combined 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 Oregon.
SIP is usually thought of as a way to implement the functionality of common telephony over an IP network. It is replacing the older, less flexible protocols used in the old days such as H.323 and MGCP. These earlier protocols performed at a very low level to connect IP phones to the public telephone system. SIP, however, provides an elegant and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard phone network.
SIP trunking provides many perks to the business user such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the huge subscription expenses of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by providing both voice and data over the same connection. Businesses will have the flexibility to route calls to favored carriers and the redundancy of using several service providers.
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