Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person voice data 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 handle IP telephony combined with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Houston.
SIP is in most cases thought of as a way to provide the operation of common 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 functioned 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 instance, SIP uses email addresses as the SIP address in place of a telephone number over the standard phone network.
SIP trunking brings a number of benefits to the business person such as cost savings, networking versatility, 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 providing both voice and data over the very same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using multiple service providers.
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