Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person real time data over the Internet. 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 extended to deal with IP telephony combined with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Corinth.
SIP is commonly associated with a way to implement the functionality of ordinary telephony over an IP network. It is replacing the older, less versatile 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 network. SIP, however, 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 telephone network.
SIP trunking delivers numerous benefits to the business user such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the huge subscription costs 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 a number of service providers.
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