Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport 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 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 extended 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 Felsenthal.
SIP is in most cases associated with a technique to provide the functionality of common telephony over an IP network. It is replacing the older, less flexible protocols used in the old days such as H.323 and MGCP. These earlier protocols performed at a very low level to link IP phones to the public telephone system. SIP, on the other hand, provides an elegant 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 perks to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the huge subscription fees of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by delivering both voice and data over the same connection. Businesses will have the manageability to route calls to favored carriers and the redundancy of using multiple service providers.
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