VOIP is an acronym which stands for “Voice Over IP”. Most of us are familiar with the “Public Switched Telephone System” (PSTN), which allows us to contact people around the globe by dialling a sequence of numbers. VOIP offers an alternative, which works by routing digitized voice signals over IP networks, such as Company Intranets, or in some cases the public Internet.
On the face of it, the PSTN hasn’t really changed much in more than 100 years. There have been many technology changes and improvements, such as tone dialling and Caller ID, but as far as the user is concerned, it’s still a matter of dialling (more recently, pressing) a sequence of numbers, and getting connected to the person who’s number was dialed. However, what happens behind the scenes to make this happen has changed considerably in recent years.
VOIP isn’t a particularly new technology; there are papers and patents about the subject dating back several decades, and there was some early VOIP software available as early as 1991. The basic principle is pretty simple; it is essentially the same technology that is used to stream music across the Internet. Voice sounds are picked up by a microphone and digitized by the sound card. The digitized audio is then compressed using an audio codec. This works by removing redundant and unneeded data, while maintaining the legibility of the audio, to make the stream compact enough to be sent in real time over the network. The term codec is short for “enCODer/DECoder”. The sounds are encoded at the sending end, sent over the network and then decoded at the receiving end, where they are played back over speakers or a headset.
The only requirements are a network connection between the two computers of an adequate speed, and matching codecs at each end. Regular “off the shelf” PCs equipped with microphones, sound cards, headsets and a broadband connection fit the bill perfectly.
SIP, or Session Initiation Protocol, is a signalling protocol, which is widely used for setting up and tearing down multimedia communication sessions, such as voice and video calls over IP. Other potential application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. SIP is a request-response protocol, which closely resembles two other Internet protocols, HTTP and SMTP, the protocols underlying the World Wide Web and email, respectively.
SIP provides four basic functions:
- Translation from a user’s name, to their current network address
- A mechanism for call management – adding, transferring and dropping participants.
- Provides feature negotiation, so that all participants can agree on the features to be supported.
- Allows for changes to the supported features during a call.
The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions, consisting of one or more media streams. The simplicity of the specification means that SIP can scale, it is extensible, and it sits comfortably with different architectures and deployment environments. Although SIP was developed as a mechanism to establish sessions, it does not need to know the details of a session; it just initiates, terminates and modifies sessions.
Several other VoIP signalling protocols exist, but SIP is distinguished by its proponents for having roots in the IP community rather than the telecom industry. SIP has been standardized and governed primarily by the IETF while the H.323 VoIP protocol has been traditionally more associated with the ITU. However, the two organizations have endorsed both protocols in some fashion.
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