Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport technology for person-to-person voice traffic over the Web. Its defining 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 combined with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Bloomingdale.
SIP is in most cases thought of as a means to provide the functionality of standard telephony over an IP network. It is replacing the older, less versatile protocols used in the past such as H.323 and MGCP. These earlier protocols worked at a very low level to link IP phones to the public telephone system. SIP, on the other hand, provides a sophisticated and straightforward 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 brings multiple advantages to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the high 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 flexibility to route calls to preferred carriers and the redundancy of using several service providers.
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