Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications protocol for person-to-person real time data over the Web. Its technical 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 allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Hebo.
SIP is commonly thought of as a way to emulate the functionality of ordinary telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These older protocols performed at a very low level to connect IP phones to the public telephone network. 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 regular telephone network.
SIP trunking brings numerous perks to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the high subscription costs 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 manageability to route calls to preferred carriers and the redundancy of using multiple service providers.
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