Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person voice traffic over the Web. Its defining 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 handle IP telephony combined with video and instant messaging. Forthcoming enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Jackson.
SIP is typically associated with a technique to implement the operation of common 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 older protocols worked 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 instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard phone network.
SIP trunking provides numerous benefits to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the huge subscription fees 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 preferred carriers and the redundancy of using multiple service providers.
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