Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications technology for person-to-person voice data 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 handle IP telephony combined with video and instant messaging. Forthcoming improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Halls.
SIP is typically thought of as a means to emulate the operation of typical telephony over an IP network. It is replacing the older, less adaptable protocols used in the old days such as H.323 and MGCP. These older protocols performed at a very low level to connect IP phones to the public telephone network. SIP, on the other hand, 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 delivers multiple advantages to the business person such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the high subscription fees of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by delivering both voice and data over the very 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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