Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology for person-to-person voice traffic over the Internet. 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 handle IP telephony in conjunction 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 Marlborough.
SIP is typically associated with a method to provide the operation of ordinary telephony over an IP network. It is replacing the older, less adaptable protocols used in the past such as H.323 and MGCP. These earlier protocols performed at a very low level to connect IP phones to the public telephone system. SIP, however, 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 regular telephone network.
SIP trunking delivers numerous advantages to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the high subscription expenses of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by providing 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 multiple service providers.
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