Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person voice 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 extended to handle IP telephony in conjunction 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 Scott City.
SIP is usually associated with a method to emulate the operation of common telephony over an IP network. It is replacing the older, less versatile protocols used in the past such as H.323 and MGCP. These earlier protocols functioned 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 instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking provides many advantages to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the huge subscription expenses 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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