Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice data over the Internet. Its technical 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 improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Gordon.
SIP is commonly thought of as a means 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 earlier protocols functioned at a very low level to link IP phones to the public telephone network. SIP, however, provides a sophisticated and straightforward way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard telephone network.
SIP trunking brings many perks to the business person such as cost savings, networking versatility, 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 providing both voice and data over the same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using several service providers.
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