Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications protocol for person-to-person voice traffic over the Internet. Its technical 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 expanded to handle IP telephony in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Lebanon.
SIP is usually thought of as a technique to emulate the functionality of ordinary 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 worked at a very low level to link IP phones to the public telephone network. SIP, however, provides a sophisticated and straightforward way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the regular telephone network.
SIP trunking provides multiple advantages to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate 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 manageability to route calls to preferred carriers and the redundancy of using multiple service providers.
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