Are you willing to work in Networking sector? Are you a graduate in computer engineering/IT/Electronic engineering then logon to Wisdom jobs online site. VoIP telephony system uses packet switched voice over IP or Internet telephony to transmit telephone calls over the Internet as opposed to the circuit switched telephony used by the traditional Public Switched Telephone Network (PSTN). It is a group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol networks. It is a category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls by sending voice data in packets. So place yourself as Network or Voice Engineer Unified Communication Engineer, System Engineer, IP Telephony Engineer by looking into VoIP job interview question and answers given.
VoIP stands for Voice over Internet Protocol or Voice Over IP. VoIP technology makes it possible to convert analog voice signal into digital data and transmits it over the Internet. (There are more likely possible pronunciations, as well as vo-ipp, have been used, but generally, the single syllable - VoIP, as in voice - may be the most common within the industry.)
In addition to its cost efficiency, the feature-rich services and the metaphorical disappearance of geographical boundaries as that were mentioned above; VoIP has many other benefits as follows:
VoIP is usually based on the SIP system that is the recognized standard. Any SIP compatible device can talk to any other. Any SIP telephone can call another over the Internet - any additional equipment is not needed for this. You only need to plug your SIP phone into the Internet connection, configure it then dial the other person. You can also connect traditional analog phones to your VoIP network – in this case and ATA device is needed.
In VoIP systems, your analog voice is converted into packets of data (as little files), and then transmitted to the recipient through the Internet and decoded back into your voice at the other end. To make it quicker, these packets are compressed before transmission, a bit like zipping a file (it will be decompressed of course at the other party).
The advantages of converting analog signals into digital data can be summarized as follows: Digital format can be better controlled as it can be compressed, routed, converted, etc. In addition, digital signals are more noise tolerant than analog signals. Quality of Service (QoS) ensures real-time errorless data streaming that allows interactive data voice exchange as well.
If you only want to use VoIP to communicate with other users in your VoIP network, you can do that free of charge. If however you want to be able to use VoIP to make and receive calls to/from people who are out of your VoIP network or do not have VoIP, you will need to subscribe to a VoIP service provider plan, and a gateway service may be also needed that provides a bridge between VoIP and the conventional phone networks.
Definitely yes. VoIP is a very cost-effective option for those companies who want to upgrade their old PBX systems and VoIP ensures new features that traditional PBX systems simply do not. To change to a VoIP system, companies can buy an IP PBX, but it is also possible to add some VoIP functionalities into an existing phone system.
Getting started with VoIP is fairly simple. Assuming that you already have the 2 most important ingredients (a Windows PC or Mac computer and a broadband Internet connection), all you need to get started is the following:
You can use a headset of course rather than a microphone and speaker to leave your hands free.
In order to choose which software to use, it is worth to consider the followings. Using voice chat in Talk or Yahoo Messenger could be regarded as VoIP, so could the highly publicized Skype; but these are all proprietary systems. You can download them free of charge, but to talk to someone using Talk, the person at the other end also needs G-Talk. The same applies to Yahoo and, to a great extent, to Skype. They use their own special system that is not open and will not connect to other systems easily. So – especially for corporate using – it is rather recommended to use such a soft phone as Counterpart X-Lite with a SIP enabled IP PBX or access to an Internet Service Provider.
VoIP has been implemented in various ways using both proprietary protocols and protocols based on open standards. You can see the VoIP protocols below:
The most commonly used one is SIP. Session Initiation Protocol is IETF signaling protocol used for VOIP and other text and multimedia communication sessions such as voice and video calls over Internet Protocol (IP).
SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. These sessions include Internet telephone calls, multimedia distribution, multimedia conferences, instant messaging, file transfer and online games.
The best way for creating any VoIP application is using a VoIP development kit. These SDKs are intended to provide a background support for your VoIP project by offering prewritten VoIP components. It is quite effective and comfortable to use these prewritten components, as you can save time and money as well. (During the soft phone development that will be described below, Osaka VoIP SIP SDK has been used for this purpose, that supports all the .NET programming languages, so C# as well.)
For using these toolkits, you have to add your preferred SDK as reference in you IDE. After you have added it, you can reach all VoIP components that are needed to be able to define the behaviour of such VoIP applications as soft phones, call recorders, IVR menu systems, software-based IP phone systems (PBX), etc.
In order to be able to make voice calls by using your own software application you need to connect your system to the telephone network.
This can be done in three ways:
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