Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission 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 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 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 companies in Walcott.
SIP is in most cases thought of as a means to provide the operation 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 older protocols functioned at a very low level to connect IP phones to the public telephone system. SIP, on the other hand, provides an elegant and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address in place of a telephone number over the standard telephone network.
SIP trunking provides many advantages to the business person 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 same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using several service providers.
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