Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission technology for person-to-person voice data over the Web. Its technical 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 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 Lyon.
SIP is in most cases associated with a technique to emulate the operation of standard telephony over an IP network. It is replacing the older, less versatile protocols used in the old days 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, on the other hand, provides a sophisticated and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking provides a number of perks to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the high subscription fees 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 manageability to route calls to favored carriers and the redundancy of using multiple service providers.
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