1. VOIP Introduction and Overview Voice over IP (VOIP) is arguably one of the most important developments in the field of telecommunication and networking. VOIP allows us to transmit telephone conversations or voice as data packets over the Internet Protocol (IP); as a result VOIP can be implemented of any data networks that would use IP. Before transmitting the voice over IP network it is digitized and converted to IP packets. Since the internet is freely available throughout the world now it is possible to use VOIP in a higher degree. Let’s discuss in detail what VOIP actually is and how it works. We will also discuss the different protocols used to implement VOIP and finally we will discuss the benefits and disadvantage of VOIP. What is VOIP? VoIP is also known as Internet Telephony or IP Telephony, VOIP allows us to make telephone calls over the Internet using a broadband internet connection instead of the regular analog telephone lines. VOIP converts the voice signal from our telephones into digital signals which travel over the Internet. We can use either a telephone or a PC as a user terminal while placing calls with VOIP. Hence we can make either PC to PC call, PC to telephone calls or telephone to telephone calls with VOIP via the internet. (Voice over IP – By Mark Leppänen) One of the main reasons why VOIP has become so popular with individual consumers is that one can make long distance phone calls over the internet thereby bypassing the toll charges they would have normally paid over traditional telephone networks. Since VOIP allows us to integrate voice and data; organizations are heavily inclined to carry their voice applications over the already existing data networks thereby reducing the overall maintenance cost and saving a lot of money. However, as the usage of VOIP increases we need to be aware of the threats and vulnerabilities associated with using VOIP; these threats are similar to what a user would experience while using the internet. According to the author of the paper ‘Understanding VOIP’; various new scams and threats unique to IP telephony are emerging with the rise in use of VOIP. (Understanding VOIP – By Matthew Desantis, US-CERT, 2008) How does VOIP work? When we record our voice using a microphone we are basically sampling the voice and storing the recorded voice in a video tape, CD etc. With VOIP we are basically doing the same thing, VOIP converts our voice into digital signals which are compressed and placed into data packets and transmitted over the Internet. The process of placing digital voice signals in data packets is called as 'packetization'. It is possible that while the packets are being transmitted over the Internet they can get delayed or lost. VOIP has several mechanisms to avoid packet loss and packet delay. Once the packet is successfully received at the destination it is then decompressed and converted into analog signals. To ensure successful sending and receiving of packets over the internet it is important that the devices that take part in VOIP communications follow a protocol to identify each other. We also need a protocol which ensures how information is exchanged between the communicating devices. We can implement VOIP using a variety of devices like IP phone, Computers, Analog Terminal Adapters etc. Let us understand various VOIP protocols in the next section. 2. VOIP Protocols Many protocols have been developed in order to develop VOIP services. Some of these protocols are based on open standards while some are proprietary protocols. We will focus on the most popular protocols that are being used to implement VOIP services. The sections below give a basic overview of each of these protocols. RTP RTP stands for Real-Time Protocol; this standard is implemented by almost every device to transmit audio and video packets over the internet. RTP is typically implemented on top of UDP. At the sender's side the media is encapsulate in an RTP packet which in turn is encapsulated in the UDP segment. The receiver extracts the RTP packet first from the UDP segment and finally the media chunk in extracted. The RTP packet has four main packet header fields namely Payload Type, Sequence Number, Timestamp and Source Identifier field. They 'Payload Type' field as the name suggest gives us information about the type of encoding used for the media. The sequence number field allows the receiver to keep track of the packets and helps the receiver to identify packet loss. The timestamp field allows the receiver to have a synchronous play out of the media and it also allows the receiver to take care of the packet jitter introduced by the network. The 'Source Identifier' field helps the receiver identify the originator of the media frame; it is very helpful in a multicast session where there would be many participants. (Computer Networking: A Top-Down Approach; James F. Kurose and Keith W. Ross, Pearson Education, 2013) H.323 The H.323 protocol is a st