Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person real time data over the Internet. 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 extended to handle IP telephony in conjunction with video and instant messaging. Future enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Brunswick.
SIP is typically thought of as a method to emulate the functionality of common 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 earlier protocols performed 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 in place of a telephone number over the standard phone network.
SIP trunking delivers numerous advantages to the business person such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with 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 flexibility to route calls to preferred carriers and the redundancy of using a number of service providers.
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