Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transport 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 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 combined with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Sugar Hill.
SIP is commonly thought of as a way to implement the functionality of standard 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 performed at a very low level to connect IP phones to the public telephone network. 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 regular phone network.
SIP trunking delivers many benefits to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the huge subscription expenses 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 a number of service providers.
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