Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications 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 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 such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in South Hutchinson.
SIP is in most cases thought of as a technique to emulate the functionality of standard telephony over an IP network. It is replacing the older, less adaptable protocols used in the old days such as H.323 and MGCP. These earlier protocols worked at a very low level to connect 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 instead of a telephone number over the regular phone network.
SIP trunking delivers a number of perks to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the high subscription expenses 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 favored carriers and the redundancy of using multiple service providers.
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