Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) 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 extended to deal with IP telephony combined with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Waldron.
SIP is commonly thought of as a means to emulate the functionality of standard telephony over an IP network. It is replacing the older, less flexible protocols used in the old days such as H.323 and MGCP. These earlier protocols worked at a very low level to link IP phones to the public telephone system. SIP, however, provides a sophisticated and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address in place of a telephone number over the regular phone network.
SIP trunking provides multiple advantages to the business user 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 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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