Why You Need SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transmission 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 extended to deal with IP telephony combined with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Princeton.
SIP is commonly associated with a technique to provide the operation of typical 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 worked at a very low level to connect IP phones to the public telephone network. SIP, however, provides a sophisticated and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the regular phone network.
SIP trunking brings many benefits to the business user such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the high subscription expenses 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 manageability to route calls to preferred carriers and the redundancy of using several service providers.
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