Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person voice traffic over the Web. 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 expanded to deal with IP telephony combined with video and instant messaging. Forthcoming improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Fulton.
SIP is typically thought of as a way to emulate 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, however, provides an elegant and straightforward way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the regular phone network.
SIP trunking brings a number of advantages to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the huge 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 multiple service providers.
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