Voice over IP (VoIP)

Author: CCa2z

Date: 12th November 2004

VoIP is 'using data network technologies and infrastructures as a medium for carrying real time voice communications' - this also applies for video over IP. Converting speech into data and transporting that data across a network, converting it back into speech at the other end means instead of using telephone lines to carry the voice it is possible to use a data network.

How does it work?

The majority of voice traffic has been digital for some time, there is nothing new in converting analogue sound into a digital form for transmission. The conversion is achieved by sampling the sound wave at a regular and rapid interval resulting in a stream of values, or bit-stream. The bit-stream is produced by a CODer-DECoder, or CODEC for short. The CODEC function can employ an algorithm to compress the resulting bit-stream and therefore reduce the bandwidth needed to transmit the voice. Bandwidth is simply a measure of how fast the ones and zeros are produced from the CODEC, or to put it another way - a measure of how many ones and zeros are produced per second.

In traditional digital telephony such as that used in the public network the standard CODEC is called G.711: this produces a 64kbps (64,000 bits per second) bit-stream. Although this is quite a high bandwidth figure, it does provide a very high quality voice service. This CODEC can also be used with VoIP networks or if bandwidth is tight the more modern and efficient G.729 algorithm can be used, producing a bit-stream of 8kbps; with only a slight drop in voice quality.

So, now that the voice sound wave has been converted into a bit-stream, how is it transmitted across a data network?

In order for this bit-stream to be sent over a data network it must exist amongst the existing data network traffic, therefore conforming to the same language or protocol as the rest of the data traffic of TCP/IP protocol (IP). The bit-stream is broken down into small slicesp acketisation or encapsulation, and an IP header is introduced - this can turn an 8kbps bit-stream into a 22kbps bit-stream. This is not all bad news however as mechanisms exist to reduce this in some cases. The data network equipment uses the information in the IP header to make routing decisions and also priority decisions.

Characteristics of networks, data and voice packets

The characteristics of networks are as follows.

  • Bandwidth - how many bits can be transmitted per second
  • Latency - how long does it take to get from one place to another
  • Jitter - the variations in latency over time
  • Packet loss - are packets lost in transit, and if so how many per second
  • Packet size - the length of the packet in bytes

Within these categories traditional data packets have the following characteristics

  • Bandwidth - variable and bursty by nature
  • Latency - tolerant of high latency for most applications but not all
  • Jitter - tolerant of high jitter for most applications but not all
  • Packet loss - intolerant of packet loss and requires retransmission mechanisms within the protocol
  • Packet size - data packets vary in size

Within these categories voice packets have the following characteristics

  • Bandwidth - constant bit-rate
  • Latency - intolerant of high latency (up to 150ms)
  • Jitter - intolerant of high jitter
  • Packet loss - tolerant of packet loss
  • Packet size - constant bit-rate

Quality of Service (QoS)

Basically voice bit-streams are what are referred to as 'real-time-data'. That is to say it must be played back at the far end at a constant rate without interruptions or disruptions. If not it will either result in poor sound quality or be completely unintelligible. The above characteristics hint at the problem faced when running voice packets over a data network. Whilst voice bit-streams are constant and fairly low in bandwidth; data is bursty and unpredictable in nature. Therefore particularly in a low bandwidth network or link there is the risk of data traffic bursts getting in the way of the voice bit-streams and causing delays and variations in sound quality.

It is critical to deploy measures that will maintain the transmission characteristics of the voice packets within an acceptable level. These measures are called quality of service measures (QoS). QoS is the science of configuring the data network equipment to maintain voice quality by protecting the voice bit-streams from unpredictable traffic patters of the other data traffic.

The success of a VoIP deployment, relies upon getting this right and providing a management framework over the network that will maintain this (basically strong procedures, change control, and general best practice). A welcome side-effect of correctly deploying VoIP onto a data network is the raised standards it can bring to the data network infrastructure in the form of stronger procedures and better infrastructure design. This often results in improved service for all network users and services.

Typical Uses

VoIP is ideal for companies with large data networks that make numerous telephone calls between sites. The existing data networks of these companies can be used to carry the voice calls and save on telephone charges. This is often referred to as 'toll bypass'.

Some companies have already built there own separate telephone networks by leasing circuits between sites for the purpose of carrying their voice and bypassing these tolls. However this means they are paying for two networks, one for the voice and one for the data. By using the data network to carry the voice they can do away with the separate voice network and reduce costs even further.

One of the reasons this can be done is VoIP uses less bandwidth than traditional voice carrying methods. This is because during the process of converting the voice into data it is compressed by up to five times ( G.729 CODEC ). Furthermore, the conversion process can recognise silence between the speech breaks and does not send this silence, reducing bandwidth requirements still further.

Business Benefits

  • Cost reduction - the integration of voice, fax, video and data can offer significant savings.
  • Reduction in bandwidth required to deliver the voice traffic - traditional telephony requires 64kbps to transport a single voice call; VoIP can use as little as 12kbps. For many companies, there is sufficient reserve capacity on data networks to transport considerable voice traffic, making voice essentially free.
  • Simplified management - remote management of the entire voice and data system becomes increasingly easy. Increased standardisation of network equipment and protocols across the network, effectively reduces the amount of networking equipment required.
    (Sabio)

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