Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time data 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 deal with IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Warrior.
SIP is in most cases thought of as a means to implement the functionality of common 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 network. 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 perks to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the huge 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 flexibility to route calls to favored carriers and the redundancy of using a number of service providers.
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