Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol 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 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 improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Raleigh.
SIP is usually thought of as a technique to provide the operation of ordinary telephony over an IP network. It is replacing the older, less adaptable 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, however, provides a sophisticated 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 delivers many benefits to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the huge subscription expenses of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by delivering both voice and data over the same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using a number of service providers.
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