VOIP and SIP News

With all the talk of VoIP takeovers and business manoeuvres this week (I am specifically thinking of Cisco's rumoured move to acquire Skype), it is easy to lose sight of the actual marvel that we have this technology.

Now I am not normally a person who looks at something and thinks 'how does that work', but I have found myself wondering about the mechanics of VoIP this week.

So when you speak into your computer microphone next time, have a think about this.

Each individual sound in your voice is broken down and recorded as a unit of data which is then assigned a numerical value. This is done with an Analogue-to-Digital Converter (ADC).

Right now your voice has been digitalised. The next step is to compress the sound down into an audio file using a codec. The data is reduced, but quality is not compromised.

This data stream is broken into smaller packets, but each are assigned a sort of address that allows them to be reconstructed in the right sequence. In this respect it is a bit similar to the technology used to send an email.

However the transport layer used by VoIP is called User Datagram Protocol (UDP) with an application layer known as Real-time Transmission Protocol (RTP). This is an optimum method of delivering audio and video over the Internet.

As well as recognizing packets of information, the RTP sometimes drops them if they have not arrived quickly enough to be sequenced. If it didn't there would be long delays in speech. That is why it is essential to have a fast internet connection. If you do not you will hear cut offs in the conversation.

Once the voice data has arrived, it is reorganized and converted from digital to analogue so that your call is comprehensible.

Pretty amazing huh?