Hello everyone, this is my debut towards Voip. I’m jotting down the points to remember so that it’s beneficial for me and for anyone intreseted in it.  Let’s go on…

VoIP is a Layer 3 network protocol that uses various Layer 2 point-to-point or link-layer protocols such as PPP, Frame Relay, or ATM for its transport. VoIP enables Cisco routers, access servers, and multiservice access concentrators to carry and send voice and fax traffic over an IP network. In VoIP, digital signal processors (DSPs) segment the voice signal into frames and store them in voice packets. These voice packets are transported via IP in compliance with a voice communications protocol or standard such as H.323, Media Gateway Control Protocol (MGCP), or Session Initiation Protocol (SIP).

SIP is used to create and control the communications sessions that are the basis of VoIP telephony. SIP is also used for instant messaging, presence updates, multimedia, conferencing, and other real-time services that can traverse a SIP trunk. VoIP protocols typically use Real-time Transport Protocol (RTP) for the media stream or speech path. RTP uses User Datagram Protocol (UDP) as its transport protocol. Voice signaling traffic often uses Transmission Control Protocol (TCP) as its transport medium.

Three separate problems to solve before making a voIP calls:
Alerting someone that you want to call them
Turning voice into digital sound and sending it over the Net (and receiving replies in the opposite direction),
“interfacing with” (linking in to) the ordinary telephone network, if your call is going to a traditional landline telephone or cellphone(mobile phone).

Call signaling

The two protocols that cover signalling are technically known as H.323 and SIP (Session Initiation Protocol, sometimes also known as RFC 4168). Simply speaking, these protocols set up a communication route between two IP addresses (the sender’s and the receiver’s) across which the actual telephone call data can be sent and received.

Call Transmission

To send a basic telephone call over the Internet, you have to turn a speaker’s voice into digital (numeric) form. The piece of software responsible for this process—converting audio sound into digital data and back again at the other end—is known as a CODEC (Coder-Decoder).  The data-sending protocol used in VoIP is called RTP (real-time protocol). Once a spoken voice has been turned into numbers, it’s relatively easy to break it into packets and send it over the Internet to another computer, where it can be reassembled and turned back into the sound of a voice by exactly the reverse process. Again, the computers involved in sending and receiving the data have to work according to the same protocols (agreed methods).

Interfacing with the telephone network.

Here comes the gateway, which acts as a bridge between the Internet (on one hand) and the PSTN (on the other). You can think of a gateway as a kind of translator that converts telephone calls in IP-format into traditional signals that ordinary phones can understand (and vice versa). It’s also involved in call signaling, so when you dial a landline from a VoIP phone, the gateway converts the call-signaling data into a format that the PSTN can understand (and rings the landline the old-fashioned way).

That’s it for this post and lets see something new in the next post.