Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport protocol 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 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. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Allen.
SIP is in most cases associated with a technique to implement the functionality of ordinary telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These earlier protocols functioned at a very low level to connect 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 standard phone network.
SIP trunking delivers a number of benefits to the business person 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 providing both voice and data over the same connection. Businesses will have the flexibility to route calls to preferred carriers and the redundancy of using several service providers.
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