Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person voice traffic over the Internet. 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 expanded to handle IP telephony combined with video and instant messaging. Future improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Hudson.
SIP is typically associated with a way to emulate the operation of common telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These earlier protocols performed 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 instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking brings multiple benefits to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate 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 very same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using multiple service providers.
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