Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport technology for person-to-person real time 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 expanded to handle IP telephony combined with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Lyon.
SIP is in most cases associated with a means to implement the functionality of standard telephony over an IP network. It is replacing the older, less adaptable protocols used in the old days 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 a sophisticated 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 regular telephone network.
SIP trunking brings numerous perks to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can do away with 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 several service providers.
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