Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transport technology for person-to-person voice data over the Web. 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. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Enterprise.
SIP is typically thought of as 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 functioned at a very low level to link IP phones to the public telephone network. SIP, however, provides an elegant and straightforward 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 provides multiple benefits to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate 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 very same connection. Businesses will have the flexibility to route calls to preferred carriers and the redundancy of using multiple service providers.
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