Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the communications protocol 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 extended to handle IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Port St Joe.
SIP is commonly thought of as a means to emulate the functionality of standard telephony over an IP network. It is replacing the older, less adaptable protocols used in the past such as H.323 and MGCP. These earlier 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 instance, SIP uses email addresses as the SIP address instead of a telephone number over the regular phone network.
SIP trunking brings a number of perks to the business person such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the huge 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 manageability to route calls to favored carriers and the redundancy of using several service providers.
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