Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transport protocol for person-to-person voice data over the Internet. 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 in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Gilmore.
SIP is usually thought of as a way to emulate the functionality of ordinary telephony over an IP network. It is replacing the older, less adaptable protocols used in the past such as H.323 and MGCP. These older protocols performed at a very low level to connect IP phones to the public telephone network. 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 in place of a telephone number over the standard telephone network.
SIP trunking brings numerous perks 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 providing both voice and data over the same connection. Businesses will have the manageability to route calls to favored carriers and the redundancy of using multiple service providers.
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