Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person real time data 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 extended to handle 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 businesses in Bergman.
SIP is typically associated with a way to provide the operation of common 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 link IP phones to the public telephone system. SIP, on the other hand, 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 a number of perks to the business person such as cost savings, networking versatility, 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 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 several service providers.
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