Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice traffic over the Web. Its defining 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 like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Monroe.
SIP is commonly thought of as a way to emulate the operation of ordinary 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 earlier protocols functioned at a very low level to connect IP phones to the public telephone network. SIP, however, 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 regular telephone network.
SIP trunking brings multiple benefits 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 delivering both voice and data over the very same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using a number of service providers.
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