Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice traffic over the Web. Its technical 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 expanded to deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Star City.
SIP is commonly thought of as a method to emulate the operation of common 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 older protocols worked at a very low level to link IP phones to the public telephone network. SIP, on the other hand, 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 delivers multiple advantages to the business person such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the huge subscription costs 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 manageability to route calls to preferred carriers and the redundancy of using a number of service providers.
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