SIP FAQ & INFO
As internet usage evolves, real-time person to person communication has become the next landmark in internet communication. SIP is the internet standard for such communication.
SIP (Session Initiation Protocol) was developed by the IETF and published as RFC 3261.
SIP is an internet protocol for live communications used in setting up and tearing down voice or video calls. It is a signaling protocol used to create, modify, and terminate sessions with one or more participants in an IP network. A session can be a straightforward two-way phone call or it can be a multi-media conference session with many persons participating.
SIP has made possible an array of services that seemed unthinkable just a few years ago: internet conferencing, IP telephony, instant messaging, presence, voice and video communication, data collaboration, online gaming, application sharing, and much more.
SIP is doing for real-time communications what HTTP did for the web and SMTP for email. It is the main driver in the acceleration of the IP Telephony revolution.
The SIP protocol is similar to the HTTP protocol as it is text based, very open and flexible. It has, as a result, largely taken over the H323 standard.
With SIP Telephony, a viable alternative to traditional PBX has emerged. SIP telephone systems deliver features that enhance users’ mobility and productivity, while securing substantial cost-saving advantages. This is making proprietary hardware based PBXs obsolete.
Some of the most common questions about SIP are answered in this SIP FAQ.
IP PBX, SIP & VOIP common questions 
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