Why You Need SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transport protocol 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 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 in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Freeland.
SIP is typically thought of as a way to implement the functionality 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 older protocols worked at a very low level to connect IP phones to the public telephone network. SIP, on the other hand, provides an elegant and well-defined way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking brings numerous advantages to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the high subscription expenses of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by providing both voice and data over the same connection. Businesses will have the flexibility to route calls to favored carriers and the redundancy of using a number of service providers.
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