Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person real time traffic over the Internet. 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 expanded to handle IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Huntington Park.
SIP is typically associated with a means to implement the operation of ordinary telephony over an IP network. It is replacing the older, less flexible 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 a sophisticated and straightforward way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the regular phone network.
SIP trunking brings numerous benefits to the business user such as cost savings, networking versatility, and emergency disaster recovery. It can eliminate the huge 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 preferred carriers and the redundancy of using several service providers.
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