VoIP stands for Voice over Internet Protocol. This phenomenon has made a profound change in the world of telephone communications. The traditional method of making calls the landlines are being fats replaced by this technology that has taken the world by storm. Not only is this method economical as this does not involve the telephony company charges that are pretty heavy, it also gives you better coverage.

This works through the broadband and uses the World Wide Web to route its calls. The best thing about it is that the person who you are calling through the internet phone need not necessarily have it also to receive your call. A technology that has brought a revolution in our worlds at least deserves that we look into its history.

Voice over Internet Protocol, VoIP or Broadband phone service as it is often referred to, is changing the telephony world. Traditional phone lines are slowly being phased out as businesses and households around the world embrace the benefits and features that VoIP technology has to offer.

VoIP is, in simple terms, the process of breaking up voice/audio into small chunks (or packets), compressing the chunks, transmitting those chunks over an IP network (e.g. the Internet), and reassembling those chunks at the receiving end, so that two people can communicate using voice/audio.

The history of VoIP shows that this technology started as far back as 1995 when a small company called Vocaltec released, what was believed to be, the first internet phone software. This software was designed to run on a home PC and much like the PC phones used today, it utilized sound cards, microphones and speakers. The software was called “Internet Phone” and used the H.323 protocol instead of the SIP protocol that is more prevalent today. Vocaltec had initial success with Internet Phone, and had a successful IPO in 1996. It was the Skype of the mid 90s. A major drawback in 1995 was the lack of broadband availability, and as such, this software used modems which resulted in poor voice quality when compared to a normal telephone call.

By 1998, VoIP traffic had grown to represent approximately 1% of all voice traffic in the United States. Entrepreneurs were jumping on the bandwagon and were creating devices which enabled PC-to-phone and phone-to-phone communication. Networking manufacturers such as Cisco and Lucent introduced equipment that could route and switch the VoIP traffic and as a result by the year 2000, VoIP traffic accounted for more than 3% of all voice traffic.

PSTN and mobile network providers

It is becoming increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks to connect switching stations and to interconnect with other telephony network providers (this is often referred to as ‘IP backhaul’).

Many telecommunications companies are looking at the IP Multimedia Subsystem (IMS) which will merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable them to upgrade their existing systems while embracing Internet technologies such as the Web, email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems to interface with the conventional PSTN and mobile phone networks.

“Dual mode” telephone sets, which allow for the seamless handover between a cellular network and a Wi-Fi network, are expected to help VoIP become more popular.

Phones such as the NEC N900iL, many of the Nokia Eseries and several other Wi-Fi enabled mobile phones have SIP clients built into the firmware. Such clients operate independently of the mobile phone network (however some operators choose to remove the client from subsidised handsets). Some operators such as Vodafone actively try to block VoIP traffic from their network. Others, like T-Mobile, have refused to interconnect with VoIP-enabled networks as was seen in the legal case between T-Mobile and Truphone, which ultimately was settled in the UK High Court in favour of the VoIP carrier.

VoIP can facilitate tasks and provide services that may be more difficult to implement using the PSTN. Examples include:

The ability to transmit more than one telephone call over the same broadband connection. This can make VoIP a simple way to add an extra telephone line to a home or office.

Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone connection over traditional phone lines, like digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.

Location independence. Only an Internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.

Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books, and passing information about whether others (e.g., friends or colleagues) are available to interested parties.

Because the underlying IP network is inherently unreliable, in contrast to the circuit-switched public telephone network, and does not inherently provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP implementations face problems mitigating latency and jitter.

Voice travels over IP networks in packets in the same manner as data, so when you talk over an IP network your conversation is broken up into small packets. These voice and data packets travel over the same network with a fixed bandwidth. This system is more prone to congestion[citation needed] and DoS attacks than traditional circuit switched systems.

Fixed delays cannot be controlled (as they are caused by the physical distance the packets travel), however some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, DiffServ). Fixed delays are especially problematic when satellite circuits are involved, due to long round-trip propagation delay (400–600 milliseconds for links through geostationary satellites).

A cause of packet loss and delay is congestion, which can be avoided by means of teletraffic engineering.

The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing analog audio, although this further increases delay. This avoids a condition known as buffer underrun, in which the voice engine is missing audio since the next voice packet has not yet arrived. When IP packets are lost or delayed at any point in the network between VoIP users there will be a momentary dropout of voice if all packet delay and loss mechanisms cannot compensate.

It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.[citation needed]

A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (due to jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, MOS scores and R factors and configuration information related to the jitter buffer.

RFC3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signaling protocol extensions. RFC3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.

In order to have a video conference, both the audio and video streams need to be coded and decoded. So a codec videoconference is necessary; because without it, you would not be able to view or hear the information. Mercifully, there are ways to secure free video codec, mostly from online sources.  Most downloading sites that offer music and videos will also offer codec software as well. You just may have to search around a bit. In short, a codec will transfer the information from analogue to digital and also from digital to analogue. Codec compression is necessary in video conferences because without it, the data size would remain large and would take a long time in transfer. Viewing it in real time would be difficult. A video codec is used to compress the data for digital video, whereas an audio codec will be used to compress the data for the audio parts of the transmission.

Videoconference Net exlains it well;  Codec is one of the most critical components of video conferencing because it enables the videos to be seen and the audio to be heard. If you were to not use Codec, then you would not be able to have a proper video conference because there would be no way for you to see the information that is coded. Although this all might seem quite complex, in reality it is not that difficult. Basically how a codec works is this;  it samples the signal several thousand times per second and then compresses it for transmission.

What different types of CODEC’s are available?

There are many, many different types of codec’s available but it depends on what you need in order to figure out what codec is best for you. If you are only using a web cam for your video conference, then you will have less information to code and decode but if you are having a video conference for a large boardroom of people with multiple cameras and screens, then you will need to invest in more codec’s. One of the bigger codec’s, used for larger projects is the G.711 codec. This particular codec can sample the audio at 64,000 times per second. This will be enough for most of your video conferencing needs. Another common codec is the G.729A codec. This particular one samples at a rate of 8,000 times per second. This is a very commonly used codec and guarantees good sound and video concentration.

How do you choose a codec?

Obviously there are such a wide variety of codec’s on the market, you have to make sure that you purchase or download free video codec that is right for your needs. Basically, if you look at it this way you should be fine: the more information, whether it be video or audio that you need to compress, the higher capability codec you will need. For example, playing a movie on your computer screen will require a small concentration of codec, but running an entire multi-screened video conference will require a much higher level of concentration from the codec. It is best to speak directly with the person whom you purchased your video conferencing unit from because they will most likely be able to tell you exactly what sort of codec is needed for the particular operation that you are running. Since there are so many different applications of video conferencing, one codec may be good for one aspect while it will not be sufficient for another. Even by walking into your local computer store, you will be able to have your specific questions answered quickly and efficiently. Just describe to the person what sort of set up you have for video conferencing, and if they are familiar with the technology, they will be able to point you in the right direction.

As we have learned, a codec video conference is the only way that a video conference can actually achieve its potential. Without the proper codec’s installed, there is no way that you will be able to see the video or hear the audio properly in real time, so when you purchase your video conferencing equipment, remember to keep this in mind. If you are only looking to get video and audio on your computer, then the process is simpler. If you click on ‘properties’ you can find out what kind of video and audio compression it uses, which will help you to decide which codec you need because codec’s are used to decipher methods of audio/video compression. You can also download mega packs of codec’s so that you will be prepared for any need you may have. Compression codec software will usually already be installed on most of the newer models of computers, so you will be able to access your videos and hear the audio as soon as you set up the proper codec. Video conferencing is a great tool but you need to make sure you have all the right resources in order to make sure the process is as smooth as possible.

Consideration for network access via PORTS is something you must always consider for web, VOIP, and video conferencing.

The best software is designed to use standard ports and protocols. In the event that some of the ports or protocols are blocked, the software will “fall back”, trying other standard ports.

Many web conferencing solutions make outbound connections to port 5061 on their servers (for login, presence and availability status messages, Instant Messaging (IM), meeting invitations,), 5004 for real-time data streams (audio, video, mouse pointer, Live Views, Application Sharing, etc), and port 443 and 80 for other traffic. As mentioned above, if the port or protocol is blocked, they will fall back to another port or protocol. For example, when a real-time audio stream is opened, the product client attempts to connect to the server using UDP over port 5004. If the UDP connection attempt fails, the client tries TCP over the same port. If the TCP connection fails over that port, the client tries to connect over ports 443 and 80.

Firewall and HTTP Proxy

The best web conferencing solutions work through standard firewalls as long as normal connection to the servers is allowed. Some sites may implement an HTTP Proxy for all of the Internet traffic. Hopefully your solution detects this, and will allow connection through the proxy. Hopefully your solution supports various types of authentication requests (Basic, Digest, NTLM, Kerberos,) to the HTTP Proxy.

Many web conferencing product use SIP (Session Initiation Protocol) for much of the communication between the clients and the servers. SIP was chosen because it was the emerging standard, and offered extensibility, and fit well with the many product’s architecture.

Many web conferencing solutions use User Datagram Protocol (UDP) or Transmission Control Protocol (TCP) for real-time data streams (audio, video, etc) where the loss of a data packet is not critical and retransmission of the missing data would cause unacceptable latency. For data transfers where loss of a data packet is unacceptable (file transfers, IM, Chat, etc), TCP is used.

Most solutions also use HTTP/HTTPS protocol between the client’s web browser and the servers for the browser-based access to Workspaces.