What Is Voice over IP | VoIP | Part 3
VoIP — Mass market telephony
Mass market telephony over high-speed internet connections:
Recently we have witnessed a rapid development of VoIP telephony market, and its related services, made possible by the growing popularity of fast Internet connections, also known as broadband, with subscribers sending and receiving calls in a similar manner to one with which the service was provided through the old analogue network is switched. To connect a traditional analog phone with broadband Internet connection is required interface, called ATA (Analog Telephone Adapter). Some U.S. companies use VoIP to make calls at any time within the U.S. itself, and sometimes to Canada and some European and Asian countries, using monthly rates of flat type. In any case, at present, mobile VoIP is a complement to traditional analog, rather than replace it. One of the current limitations, for example, is the inability to automatically route emergency calls.
Another challenge for these services is the proper handling of calls to the external fax machines, satellite television receivers, modems or Faxmodem, alarm dialers and other similar objects that depend on access to telephone lines for some or all of their functionality. For now, these calls succeed, but it could be completely prevented in some cases. This problem can be solved through the use of VoIP connections where the transmission speed is reduced to bits per second. If VoIP and cellular technologies will take precedence over those of today, some manufacturers will have to review and redesign their products, because they will be rendered obsolete by new technologies primarily in the United States and Canada.
Voice over IP analog and ISDN connection
The voice compression technologies dedicate a band ranging from 4 kbit / sec. to 82 kbit / sec. in less efficient compression formats such as GSM voice encryption algorithm with RPE-LTP (Regular Pulse Excitation – Long Term Prediction) LPC with loops and sampled at 13 kbit / s. There is a voice encoder at 13 kbit / s.
The human voice has a range of about 2.7 kHz, the Shannon’s theorem – also called Nyquist inequality – requires a bandwidth of at least 5.4 kHz. According to this theorem can successfully reconstruct a continuous signal varies over time, a series of discrete values when these were taken with a sampling frequency equal to or greater than twice the highest frequency contained in the input signal. For an analog signal of X Hz bandwidth corresponds to a digital signal 2 * X * N bits / sec, where N is the number of bits used to represent each sample for the item would be 4000 * 2 * 8 = 64 kbps s (there is a degree compared to 2.7 kHz), which otherwise is the same bandwidth provided by a single ISDN line.
However, with compression formats, it became possible to drastically reduce the required bandwidth, compression occurs after the digitization of voice before sending them. It is not possible to compress an analog signal.
There are also digital signal frequencies above the natural frequency of the voice, the analog signal is the limit to be met by digital sampling, the border with the sound quality better.
An analog signal has a higher overall quality of a digital and is the highest and the reference to reach. Higher frequency of digitization has limited usefulness because they do not add information to the starting signal, such as copying can not be better than the original.
For the transmission of video technology today (RealAudio and Windows Media Player) require a bandwidth of 50 kbit / sec. To avoid the flickering images, which roughly translated in Italian means flicker.
With this sampling rate for the conversion of analog voice signal into a digital you can send voice traffic over the Internet. Analog connections at speeds of 56 kbit / sec. are used only for downloading Internet pages. To download other files or downstream of heading the maximum speed is 33 kbit / sec. Also upstream of 56 K is 33.6 kbit / s.
A call over the Internet requires more bandwidth than the sampling frequency of heading both upstream and downstream, as the two-way telephone communication (full duplex).
Otherwise at least one of the two parties receives packets incorrect order and listening in practice meaningless words, syllables in the right sequence.
While data such as bits of web pages or files downloaded, the modem is able to make checks, interpolate (or rebuild) packets and bit patterns to limit damage and request a referral, there are no controls that will remedy the defects of transmission of item, the pair of modems is “invisible” to users also in the negative sense, it can not improve the quality of communication.
Using an appropriate communications protocol that voice compression format that limits sampling around 12-13 kbit / sec. is possible with a packet-voice packets transmitted per second have about 1.5 Kbytes (12,000 bits that are just, 1 byte = 8 bits), below the critical threshold that creates problems with Internet connection.
With ISDN you have a symmetric connection on two lines: the upstream speed is the same that you downstream. With a bandwidth of 64 kbit / sec. or committing even a single line, you can establish a call with Voice Over IP voice compression formats that sample less efficient at 50-60 kbit / sec. (Still referring to equal value to a quality of reproduction). By using two ISDN channels can handle two calls simultaneously Voice Over IP.
In America, as indicated above in the article, telephone operators use voice over IP to the user’s dwelling, therefore, also the last mile is digitized to allow communication. The advantage of extending the ISDN connections for digitization of the last mile is not used by European airlines.
Operationally as well as have a program like Skype or Voice Over IP Asterisk must install a voice compression codec, which is the communication protocol must be installed on the terminal. Among these codecs include: GSM 6.10, iLBC, Speex 15.2k, 8.0k Speex, G.711, G.723, G.729, G.771 A-Law, G.771 U-Law.
The latest versions of programs like Skype give a good call quality even at 56 kbps, in general, the programs that connect point to point with the PC / landline phone calls are a lot lower latency than dialing from a site Web (as for example in the service Tiscali).
Study: From Wikipedia, the free encyclopedia. The text is available under the Creative Commons.
