Understand Voice over IP Technology (VOIP) Before Implementing

It’s a world where all young and old are using internet services for performing their regular tasks. A boom in addition is Voice over Internet Protocol (Voice over IP, VoIP) which belongs to family of internet technologies, communication protocols, and transmission technologies. It enables a user to delivery voice communications and multimedia sessions over Internet Protocol (IP) networks. Therefore it benefits home as well as corporate setups voice related issues with a safety ensured.
The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission. On the other receiving side, similar steps such as reception and decoding is done which reproduces the original voice stream. VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codec’s which encode speech allowing transmission over an IP network as digital audio via an audio stream. The codec used is varied between different implementations of VoIP (and often a range of codec’s are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codec’s.

The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration. A notable proprietary implementation is the well known Skype protocol, which is in part based on the principles of Peer-to-Peer (P2P) networking.

Conference calling, IVR, call forwarding, automatic redial, and caller ID features that traditional telecommunication companies normally charge extra for, are available free of charge from open source VoIP implementations all around. Communication on the IP network is inherently less reliable in contrast to the circuit-switched public telephone network, as it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental Quality of Service guarantees. Therefore, VoIP implementations may face problems mitigating latency and jitter.

Most VoIP implementations support E.164 to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP implementations can also allow other identification techniques to be used, like Skype allows subscribers to choose Skype names whereas SIP implementations can use URIs similar to email addresses. Often VoIP implementations employ methods of translating non E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-In service provided by Skype and the ENUM service in IMS and SIP.

Support for sending faxes over VoIP implementations is still limited. Another challenge for VoIP implementations is the proper handling of outgoing calls from other telephony devices such as digital video recorders, satellite television receivers, alarm systems, conventional modems and other similar devices that depend on access to a PSTN telephone line for some or all of their functionality.