Other VoIP Systems - 1. 1 The Motivation

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1. 1 The Motivation
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bet | 5/32 | Sana | 22.07.2021 | Hajmi | 0,89 Mb. | | #15652 |
2.3 Other VoIP Systems
The main objective of this sub-section is to evaluate how other VoIP systems operate and what there constraints are, such that, a list can be drawn up of requirements to facilitate the design and implementation of the VoIPv6 application.
Common to all VoIP system are mechanisms to perform the following functions;
Record and Playback audio. Audio enters the system in analogue waves, the system needs to take these waves and transform them into a digital format for transmission and storage.
Encode and Decode audio. Digital representations of audio are large and bulky and have enormous data rates to large to be transmitted across a network in a practical fashion. So it is sensible to reduce the size of the data with the use off compression techniques normally lossy ones which remove any redundant data.
Encrypt and Decrypt sensitive data. Most VoIP systems have to protect the privacy of the user, typically using some form of encryption, where all audio data which is sent across the network is encrypted.
Combine and Split Audio Blocks into Packets. Most Systems take a number of audio blocks and pack them into packets for transmission across the network, this usually also involves adding a packet header to help the end system distinguish between packets.
Manage Call Signalling. For every VoIP system there has to be some form of signalling protocol, which deals with setting up voice calls, terminating them as well maintaining them, this may involve dynamically changing parameters such the compression ratio.
User Communication. Users like to be informed of the current of the state of the application at all times, this usually involves a graphical user interface on a pc based voice over IP solution or usually some form of LED screen on hardware based solutions.
Typically most of the currently available VoIP systems take advantage of the fact that users will accept low quality, stability issues and interoperability problems, in view of the fact that the system is free, with hardly any hardware costs if they own a PC and allows for just one access line where users can both simultaneously surf and talk. However this is not the view taken by professional VoIP developers.
Professional developers’ top aims are to resolve the stability and interoperability problems of the system, as well as keeping the quality of audio at a quality level equal or better to that offered by the traditional telephone.
Interoperability issues are usually solved within high end VoIP system with the use of gate keepers and gateways. Gate keepers primarily store the addresses of all the users in which the gate keeper is in charge for. When one VoIP application wishes to contact another they must first contact the gate keeper who either usually returns the current address of the recipient or sets up a connection with the recipient and the calling applications.
Gateways are normally also used in high end system to allow different types VoIP application or possible other telephony application to interconnect with one other, this solves a lot of the interoperability issues presented by bespoke VoIP systems.
The majority of VoIP system also actively monitors the quality of the audio during calls, such that, if a number of packets are being lost which results in a prolonged outage, the sending system is usually signalled to increase or change compression codec to reduce the congestion on the wire, and to try and remove any future outages. On the other hand if there hasn’t been a lost packet after a high number of received packets, then this usually means the network has some spare capacity. Therefore the receiving system may signal the sending system to change codec to improve the voice quality actively utilizing spare bandwidth.
Some VoIP system allow for full mobility such that instead of simply allowing for mobile terminals as such within the Mobile Phone system, where users can use their phone whilst on the move, some VoIP systems, don’t assign each terminal an address but assign each user an address to allow user roaming. This means users can both be mobile whist within a call, and also have the aspect of mobility in terms of moving from device to device. An example of this would be when an employee leaves home there contacting device would switch from their home phone to their mobile and then to their office phone when they arrive at work, this allows the user to maximise on the resources at hand since possibly at work or at home there may be extra bandwidth available in which will allow for improved quality or other services to be provided such as video.
Currently there are not many standalone VoIP applications available, the majority are intergraded into other applications such as the instant messenger, and conferencing applications which allow for other forms of communications like white board sharing and video conferencing.
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