Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transmission protocol 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 enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in De Witt.
SIP is commonly associated with a method to provide the functionality of common telephony over an IP network. It is replacing the older, less versatile 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 network. SIP, on the other hand, provides a sophisticated and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard phone network.
SIP trunking brings numerous benefits to the business person such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with 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 flexibility to route calls to preferred carriers and the redundancy of using a number of service providers.
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