Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission technology for person-to-person real time 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 handle IP telephony in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in South Point.
SIP is usually thought of as a means to implement the operation of ordinary telephony over an IP network. It is replacing the older, less adaptable protocols used in the past such as H.323 and MGCP. These older protocols performed at a very low level to link IP phones to the public telephone system. SIP, however, provides an elegant and straightforward way to network the enterprise. For example, SIP uses email addresses as the SIP address instead of a telephone number over the regular telephone network.
SIP trunking delivers numerous advantages to the business operator 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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