Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission 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 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 deal with IP telephony in conjunction with video and instant messaging. Future improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Winthrop Harbor.
SIP is in most cases thought of as a method to implement the functionality of typical telephony over an IP network. It is replacing the older, less versatile protocols used in the old days such as H.323 and MGCP. These older protocols functioned at a very low level to connect IP phones to the public telephone network. SIP, on the other hand, provides an elegant and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the regular phone network.
SIP trunking delivers multiple benefits to the business user such as cost savings, networking flexibility, 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 same connection. Businesses will have the flexibility to route calls to favored carriers and the redundancy of using a number of service providers.
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