Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport technology for person-to-person voice data 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 extended to handle IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Gypsum.
SIP is commonly associated with a way to emulate the functionality of common telephony over an IP network. It is replacing the older, less adaptable protocols used in the past such as H.323 and MGCP. These earlier protocols worked at a very low level to connect IP phones to the public telephone system. SIP, however, provides an elegant 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 provides numerous advantages to the business operator such as cost savings, networking flexibility, 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 flexibility to route calls to preferred carriers and the redundancy of using multiple service providers.
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