Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications 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 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 deal with IP telephony combined with video and instant messaging. Forthcoming improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Sanger.
SIP is in most cases 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 earlier protocols functioned at a very low level to connect IP phones to the public telephone system. 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 provides multiple advantages to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the huge subscription fees of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by providing 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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