Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person voice 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 expanded to handle IP telephony combined with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Newton Center.
SIP is usually associated with a way to implement the functionality of ordinary telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These older protocols performed at a very low level to connect IP phones to the public telephone network. SIP, however, provides a sophisticated and straightforward way to network the enterprise. For example, SIP uses email addresses as the SIP address instead of a telephone number over the regular phone network.
SIP trunking delivers multiple benefits to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the high subscription costs 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 favored carriers and the redundancy of using multiple service providers.
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