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What is VOIP Print

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What is VoIP?

VoIP stands for Voice over Internet Protocol. Both VoIP and SIP are used to govern the setup and transmission of voice calls over the internet. Take note that even though IP systems are designed to make telephone calls over the internet, they are still able to communicate with traditional voice networks as well as mobile networks.

In the traditional setting telephone calls are made by making use of circuit switching. This is where a circuit is created when you lift up the handset of your telephone to establish communication with the network, indicated by the dialling tone heard over the handset which confirms that you have contact with the public network. Once you enter the number of whomever you wish to call the public network will then transfer your call to the recipients' phone. When the recipient then answers the call a circuit is created between both parties and remains open throughout the whole of the two way communication. Once either receiver is replaced on the handset, the circuit is broken and the call ends.

In the VoIP setting, as discussed before in the section on SIP, a system of packet switching is in use. This is where your voice is broken down into small packets of data and transmitted across the internet to the recipients' phone where they are reassembled back into your voice.


What is SIP?

SIP is short for Session Initiation Protocol or simply the protocol needed to initiate a session. You may be wondering what session this is referring to and why it is necessary to have a protocol put in place to get one started. The session that we are referring to in this case is a communications session, more specifically an IP communication session between IP devices (Laptop, Voip phone) over an IP network. At its core SIP is a signalling protocol used to set up an IP communication session.

As with all communications sessions it starts with some sort of signalling as is also the case with a face to face conversation. If you want to ask one of your colleagues for some assistance on a particular matter you will signal your intent by calling their name. Once your colleague recognizes your signal to initiate a conversation they would signal back their acceptance of the signal to talk by responding: "Yes" or "How can I help". Now that your initiation signal has been accepted by your colleague the communication can commence as both parties are ready to converse.

The dialogue between you and your colleague will then take place by exchanging information in the form of sound waves. Once you have received the information you desired from your colleague you would signal the end of the conversation by thanking your colleague for the assistance and to be polite your colleague would signal back, e.g. "Thanks for your help" and "Don't mention it."

Signalling in an IP setting works on the same principal. Instead of signalling with sound waves through the air, IP devices signal with packets over an IP network. This being said there are numerous different kinds of IP packets with different functions and content, the most important thing being that both the sender and receiver of these IP packets have an understanding of the packet construct and what to do with the data it received.

In IP communications there are two types of IP packets, namely signalling packets and media packets. Signalling packets are used to establish the session and media packets which convey the audio/video. In the case of a voice session the initial analogue sound wave is encoded into 1's and 0's and sent over the IP network as a media packet. On the other side the 1's and 0's are then unpacked and decoded to reconstruct the analogue wave to be played back to the user.

Every second of conversation has many media packets that get sent over the IP network to convey the real time conversation. For a media exchange such as this to take place we are assuming that two important things have already taken place, this being that both parties know each other's location (know where media packets need to be sent to) and that both parties are making use of the same codecs to decode the content of the media packets. We are thus presented with the following two questions: How do we locate the other parties IP address and how do we decide which codec is going to be used? The need for a process that has the descriptions and rules that will allow us to locate each other and agree on codecs presents itself. In other words we are in need of a protocol.

This is where SIP comes in. SIP is essentially a rulebook that describes how to locate the other party and which codec to use. It also defines how to construct and send IP signalling packets to set up a call and how the call is to be managed.

When making a call between two SIP phones the phones can then make use of all of the rules set by SIP to construct an IP signalling packet, or a SIP packet which is populated with all of the data required to set up a call and sends it out over the IP network. On the other side the receiving SIP phone understands the rules of SIP, and recognizes the invitation to start a call will then notify the receiver of the call by ringing. After signalling has been completed and codecs have been determined, the phones then begin exchanging media packets that digitally convey the conversation. Sometime later one of the parties will signal the end of the call and both parties will stop sending media and the call is terminated.

The basic concept behind SIP is easy to understand as it is just signalling to set up and manage IP communication sessions.



The following section will contain information on how to setup VoIP for an Lations Voice service. Take note that you will not only setting up Lations Voice service on the VoIP phones that Lations are going to be offering as VoIP accounts can be configured on a desktop computer, laptop or mobile phone by making use of readily available VoIP applications. These VoIP applications can be found online or on the Google Play Store as well as the Apple App Store. This guide will contain examples of Lations Voice service configured on some common VoIP systems.




Host/Domain: sip.internet.co.za
Username **********
Password **********

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