Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person real time data over the Web. Its defining 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. Future improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Rison.
SIP is typically thought of as a method 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 functioned at a very low level to connect IP phones to the public telephone system. SIP, however, provides a sophisticated and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the standard phone network.
SIP trunking delivers numerous perks 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 same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using multiple service providers.
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