Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol 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 extended to deal with 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 Centerville.
SIP is usually thought of as a technique to implement the functionality 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 older protocols performed at a very low level to connect IP phones to the public telephone system. SIP, however, provides an elegant and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address in place of a telephone number over the standard telephone network.
SIP trunking delivers many perks to the business operator such as cost savings, networking flexibility, and emergency disaster recovery. It can eliminate 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 several service providers.
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