Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport protocol 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 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 companies in Hopkinton.
SIP is in most cases thought of as a means to provide the operation of ordinary telephony over an IP network. It is replacing the older, less versatile protocols used in the old days such as H.323 and MGCP. These earlier protocols worked at a very low level to connect IP phones to the public telephone network. SIP, however, provides a sophisticated and well-defined way to network the enterprise. For instance, SIP uses email addresses as the SIP address in place of a telephone number over the standard phone network.
SIP trunking delivers numerous benefits to the business operator 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 very same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using a number of service providers.
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