Session Initiation Protocol or SIP is a very flexible protocol that has great depth. It was designed to be a general-purpose way to set up real-time multimedia sessions between groups of participants. For example, in addition to simple telephone calls, SIP can also be used to set up video and audio multicast meetings, or instant messaging conferences. In this document, we’ll focus on SIP’s capabilities for VoIP, and how it sets up calls that then use RTP (the Real-time Transport Protocol) to actually send the voice data between phones.
SIP trunking is a Voice over Internet Protocol (VoIP) and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and Unified Communications facilities. Most Unified Communications software applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.
VoIP stands for Voice over Internet Protocol. It is also referred to as IP Telephony or Internet Telephony. It is another way of making phone calls, with the difference of making the calls cheaper or completely free. The ‘phone’ part is not always present anymore, as you can communicate without a telephone set.
VoIP has a lot of advantages over the traditional phone system. The main reason for which people are so massively turning to VoIP technology is the cost. VoIP is said to be cheap, but most people use it for free. Yes, if you have a computer with a microphone and speakers, and a good Internet connection, you can communicate using VoIP for free. This can also be possible with your mobile and home phone.