Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission technology 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 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 extended to deal with IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Rainier.
SIP is typically thought of as a means to implement the operation of ordinary telephony over an IP network. It is replacing the older, less versatile protocols used in the old days such as H.323 and MGCP. These earlier protocols functioned at a very low level to link IP phones to the public telephone system. SIP, on the other hand, provides a sophisticated 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 phone network.
SIP trunking brings multiple benefits to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the high subscription costs 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 flexibility to route calls to preferred carriers and the redundancy of using a number of service providers.
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