Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person real time data over the Web. 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 extended to deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Mountainburg.
SIP is in most cases associated with a means to implement the operation of typical telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These earlier protocols performed at a very low level to link 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 in place of a telephone number over the regular telephone network.
SIP trunking brings multiple advantages to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the high subscription fees 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 flexibility to route calls to preferred carriers and the redundancy of using multiple service providers.
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