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Voice Over IP

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VoicenData Bureau
New Update

Behind

the wallpaper of every networked office lies a myriad of cables and wires.

There’s network cabling for data, and there’s telephone wiring for voice

communication. There’s electrical wiring too, but here we’re interested

in the first two only. Efforts are underway to integrate these two, so that

voice can be transferred over your data network as well. The concept is

known as Voice over IP–or VoIP for short–and uses TCP/IP to carry voice

packets over a data network. It uses various modules to integrate your

existing telephone/PBX (Private Branch Exchange)/fax lines with your data

network.

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VoIP can be set up using what’s

called a VoIP module. This plugs into your network just like an ordinary PC.

You can then either connect regular telephones to it, or connect it to your

company’s EPABX and let everyone in the office use it. However, VoIP isn’t

feasible for intra-office communication–because why would you want to

communicate within your office through VoIP when you can directly dial? But

it could become a cost saver for inter-office communication. For example,

suppose you have offices in different cities across the country, which are

already interconnected through leased lines. You can then connect VoIP

modules at each location and use your existing WAN for inter-office voice

communication. This way, you can save on STD calls between your various

offices.

Extending this argument

further, VoIP could also be used over the Internet if you have VPN

connectivity. Here, you would have servers in different offices connected to

their respective local ISPs through static IP addresses. They would have VPN

software running and would use the Internet for inter-office data

communication. VoIP modules would connect to these servers, allowing voice

communication over the Internet as well. However, voice communication over

the Internet is currently not possible in India due to legal implications.

Inside a VoIP module

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A VoIP module

has four sub-modules that are based on two chips–a microprocessor and a

DSP (Digital Signal Processor). It has embedded software for performing all

its functions. The first sub-module–the voice packet sub-module–runs on

the DSP. This performs various voice functions like tone generation,

handling echo, compressing voice data for transmission, etc.

The other three sub-modules–the

network protocol module, the network management module, and the telephony

signaling gateway module–run on the microprocessor. The network protocol

module consists of the IP signaling stack for IP networks. In case of ATM or

Frame relay networks, it would consist of their signaling protocol stacks.

The network management module takes care of the physical interface to the

telephone jack, and generates the required signals at a level compatible

with the telephone. The telephony signaling gateway module takes care of

debouncing in case someone happens to plug in an old rotary type of

telephone. It also converts telephony signaling information into a format

compatible with the signaling protocol used by the VoIP module.

Drawbacks of VoIP

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Since voice

has to be packetized and transmitted over a data network, it raises some

problems not experienced over ordinary telephone lines. These problems

affect the overall speech quality, and include:

  • Delay: An IP

    network sends packets along the shortest route available at the time of

    transmission. So there’s a possibility of packets arriving out of

    order or being delayed. This, coupled with the time taken to re-arrange

    and re-assemble the packets, causes delay in the transmitted/received

    speech. The ITU (International Telecommunication Union) recommends that

    one-way delay should not exceed 150 ms for achieving good quality voice.



  • Jitter: One of the

    consequences of delay is jitter. This is also caused when packets are

    lost over the network. It causes the speech to sound choppy and

    difficult to understand. To counteract this, a jitter buffer is

    used, which buffers the received packets before they’re sent to the

    voice codec. A ratio of the time difference between each received packet

    is calculated and used to dynamically adjust the size of the jitter

    buffer.



  • Packet loss: This

    occurs due to degradation of packets over a network, or due to

    congestion. Continual losses of about 5-10 percent of the received

    packets can make a significant difference in voice quality. For small

    losses, the same principle is used as in your audio CD player. That is,

    the last bit of data is replayed as a substitute for the missing data.

    This usually goes unnoticed, unless a large amount of data is missing.

Currently, there are a few

solutions available for VoIP. However, the use of VoIP is still limited

because not everyone can interconnect their offices through WANs. It could

become a feasible solution if more bandwidth is available with ISPs, and the

legal implications of using voice over the Internet are taken care of.

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