Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person voice traffic 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 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 businesses in Breinigsville.
SIP is in most cases associated with a technique to provide the functionality 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 worked at a very low level to link IP phones to the public telephone network. SIP, on the other hand, provides a sophisticated and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the standard telephone network.
SIP trunking provides many perks to the business person 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 same connection. Businesses will have the manageability to route calls to favored carriers and the redundancy of using a number of service providers.
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