Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology 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 expanded to deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Lynchburg.
SIP is in most cases associated with a method to provide the operation of common telephony over an IP network. It is replacing the older, less adaptable protocols used in the past 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 example, SIP uses email addresses as the SIP address instead of a telephone number over the standard phone network.
SIP trunking provides many perks to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with 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 flexibility to route calls to favored carriers and the redundancy of using multiple service providers.
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