Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport technology for person-to-person real time data over the Internet. Its technical 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 such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Penn Yan.
SIP is commonly associated with a technique to provide the functionality of typical 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 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 regular phone network.
SIP trunking delivers a number of advantages to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the high 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 flexibility to route calls to favored carriers and the redundancy of using several service providers.
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