Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person real time traffic over the Web. Its technical 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 handle IP telephony combined with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Baxter.
SIP is commonly thought of as a method to implement the functionality of ordinary telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These older protocols worked at a very low level to connect IP phones to the public telephone system. SIP, however, provides an elegant and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address instead of a telephone number over the regular telephone network.
SIP trunking provides numerous perks to the business person such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the huge subscription costs 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 favored carriers and the redundancy of using several service providers.
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