Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport technology for person-to-person real time 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 combined with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Seymour.
SIP is commonly thought of as a means to emulate 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 older protocols performed at a very low level to link IP phones to the public telephone system. SIP, however, provides an elegant and straightforward 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 a number of perks to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the huge subscription fees 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 flexibility to route calls to preferred carriers and the redundancy of using multiple service providers.
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