Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications 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 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 combined with video and instant messaging. Future improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Gypsum.
SIP is in most cases thought of as a way to provide 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 earlier protocols worked at a very low level to link IP phones to the public telephone network. SIP, on the other hand, provides an elegant and straightforward way to network the enterprise. For example, SIP uses email addresses as the SIP address instead of a telephone number over the standard phone network.
SIP trunking provides numerous perks to the business operator such as cost savings, networking versatility, and emergency disaster recovery. It can do away with the high subscription fees 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 manageability to route calls to favored carriers and the redundancy of using several service providers.
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