Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person real time traffic over the Internet. 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 enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Mandan.
SIP is usually thought of as a means to emulate the functionality of common telephony over an IP network. It is replacing the older, less versatile protocols used in the past such as H.323 and MGCP. These earlier protocols worked at a very low level to link IP phones to the public telephone system. 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 standard phone network.
SIP trunking delivers a number of benefits to the business user such as cost savings, networking flexibility, 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 providing both voice and data over the very same connection. Businesses will have the flexibility to route calls to preferred carriers and the redundancy of using multiple service providers.
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