Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications 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 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 extended to deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Imperial.
SIP is in most cases thought of as a method to emulate the functionality of common 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 older protocols worked at a very low level to link IP phones to the public telephone network. 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 phone network.
SIP trunking brings a number of advantages to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate 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 manageability to route calls to favored carriers and the redundancy of using several service providers.
