Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time 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 extended to deal with 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 businesses in Rockdale.
SIP is usually thought of as a technique to implement the operation of standard telephony over an IP network. It is replacing the older, less versatile 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 network. SIP, however, provides a sophisticated 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 many advantages to the business user such as cost savings, networking versatility, 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 providing both voice and data over the same connection. Businesses will have the flexibility to route calls to favored carriers and the redundancy of using multiple service providers.
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