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Telephone network
When it comes to transport audio / video, there are essentially two types of problems:
-The report shows how to initiate and maintain the connection.
-Encoding indicates the manner in which to transmit the data.

Currently the telephone network, at least in the inner part, has become an IP network, based on the backbone.The most device is however still in analog form.
The telephone network is still circuit-switched.
In the telephone network there is no compression, nor is it provided communication on high quality, if not through the use of channels to multiples of 64Kbps. Does not perform silence suppression, neither the static multiplexing (you can not change the banda).
Before the start of the call phase is required for signaling (call-step).

Data network (VoIP)
In the central part also the telephone network is composed of a packet-switched network, in order to have static multiplexing, which makes it useful the silence suppression. It can also have compression. And 'possible to transmit high quality.
The data network introduces the problem of managing the quality of the call, because the band is not booked and therefore not guaranteed; also crossing times and the interarrival (jitter) are random and not guaranteed in advance.

Advantages of VoIP
Several people have seen many improvements with the arrival of VoIP:
-Consumer has no additional cost along with free wireless internet.
-Telecom can transmit more data with less bandwidth due to packet switching.
-Enterprise (business user) has no additional cost, can customize the service.

VoIP flow
It can be represented in phases:
-Sampling transforms the signal from analog to digital. E 'important to the sampling frequency and the sensitivity, which determines the quantization error. These two factors affect the theoretical bit rate.
-Encoding compresses the data. The encoding affects the actual bit rate, as it is given by the bit rate theoretical / compression factor. This stage provides for the abolition of silence (with the inclusion of white noise) and echo cancellation. The encoding requires the time and therefore introduces a certain delay (plus some coding, eg MPEG, need of subsequent frames to be calculated). The encoding can be:
- Encoding differences (without loss). It may be predictive or static.
- Weighted coding.
- Coding loss.
-Packaging is a function necessary to lower the overhead of the headers, although it introduces much delay. It improves efficiency. The packages normally contain about 20/40ms of information.
-Queuing while waiting for the packet is actually sent.Some routers try to have a queuing priority, so as to pass before packets audio / video compared to the rest of the traffic. The problem is that the task of marking the traffic is often left to the user, then you can also mark packets that are not really A / V.
-Transfer is the most delicate phase of the VoIP flow. The time needed to transmit the packet is completely bounded above by the transmission time of the packet itself. MTU is too high to have problems.
-De-jitter are used to delete the interarrival times. The packets that arrive too late are considered lost.
-Reordering
-Decoder decompresses the data. This phase is implemented by the player.
Error-checking, based on parity bits. The retransmission can not be performed, as it would require too much time.

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