Report on VoIP Call

Introduction:

Voice over IP call is used for voice communication by using internet protocol. Most popular examples of VoIP are Skype and Google Talks. Both receiving and sending ends have a unique IP addresses associated with them, the voice from sender’s end is converted into digital signals, and this digital signal is sent over IP network.

Major Devices and Components in VoIP:

VoIP phones: VoIP phones are the endpoints of the voice communication, and they provide IP addresses for each end.

GateKeeper: It provides functionalities like address translation bandwidth control and management.

GateWay: Sometimes a user may want to call to a traditional telephone number (non-VoIP call). So, in such cases, VoIP system needs to convert the voice signal into traditional telephone signal. So, this conversion is done by gateway.

MCU: MCU is acronym for Multiple Control Unit. Sometimes multiple users may want to attend a voice conference, MCU is responsible to handle the connectivity of multiple users.

Call Agent: Call agent has functionalities similar to Gatekeeper, but the major difference is that Call Agent runs on a server platform whereas Gatekeeper runs on a router.

Given diagram explains the data flow in VoIP call.

Sampling: Sampling means recording of the analog voice data. We get pulse amplitude modulation (PAM) as the output of sampling.

Quantization: For converting analog into digital, a numeric value is assigned to the signal based on amplitude of the wave.

Encoding: Each numeric value is represented using bits. For example, each sample is represented using n bits and there can be 2^n possible values of a signal. PCM (pulse code modulation) is most used strategy for converting analog voice data into digital signal. PCM takes 8000 sample per second and represents each sample using 8-bit code.

Compression: Data compression is optional, at this stage data is compressed using various strategies. So, after compression, the data is transferred using IP network to the receiving end.

Now, at the receiving end, data undergoes following stages sequentially.

Decompression: Depending upon whether data was compressed or not, data is decompressed.

Decoding: Data is decoded depending on the algorithm used for encoding.

Filtering And reconstruction of the data: Reconstruct original signal.

Quality of digital signal:

The quality of digital signal is dependent on the number of samples of voice taken per second samples of voice taken per second Sampling rate. Higher the number of samples per second better is the quality of digital signal. But as the sampling rate increases, the number of bits to be transferred per second also increases. Generally Sampling rate is equal to at least twice the highest frequency of that signal.

References:

[1] http://what-when-how.com/ccnp-ont-exam-certification-guide/digitizing-and-packetizing-voice-cisco-voip-implementations/

[2] http://what-when-how.com/cisco-voice-over-ip-cvoice/voip-fundamentals-introducing-voice-over-ip-networks-part-1/