Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time 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 combined with video and instant messaging. Future improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in South San Francisco.
SIP is typically associated with a technique to emulate the operation of typical 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 earlier protocols performed at a very low level to connect IP phones to the public telephone network. SIP, however, provides an elegant 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 phone network.
SIP trunking brings a number of benefits to the business user such as cost savings, networking flexibility, 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 very same connection. Businesses will have the flexibility to route calls to favored carriers and the redundancy of using several service providers.
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