Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications 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 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 improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Chesterfield.
SIP is usually thought of as a means to emulate 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 network. 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 standard telephone network.
SIP trunking delivers many advantages to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate 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 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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