Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice data over the Internet. 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 deal with IP telephony in conjunction with video and instant messaging. Future improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Coffee.
SIP is usually thought of as a technique to emulate the functionality of standard telephony over an IP network. It is replacing the older, less flexible protocols used in the past 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, on the other hand, provides an elegant and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking provides numerous benefits to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate the huge subscription costs 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 favored carriers and the redundancy of using several service providers.
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