Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology 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 extended to deal with IP telephony combined with video and instant messaging. Forthcoming improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Centre.
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 past such as H.323 and MGCP. These older protocols worked at a very low level to link IP phones to the public telephone network. SIP, on the other hand, provides an elegant and straightforward 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 brings numerous advantages to the business user such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the huge 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 flexibility to route calls to preferred carriers and the redundancy of using multiple service providers.
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