Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person real time 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 enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Arapahoe.
SIP is commonly associated with a technique to emulate the functionality of typical telephony over an IP network. It is replacing the older, less adaptable protocols used in the past such as H.323 and MGCP. These older protocols functioned at a very low level to connect IP phones to the public telephone network. SIP, however, provides a sophisticated and well-defined way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking brings many benefits to the business operator such as cost savings, networking flexibility, 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 providing 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 a number of service providers.
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