Voice Formant Group Travel

    Three communication fitters. The suffering of George and Harris. A victim of one hundred and seven interference. Useful recipes. The tool against diseases of the vocal tract at the installers. Installers agree that they are overworked and that they need rest. A week at sea, away from twists? George offers a river trip. Montmorency objects, cell phone catches on the river. The initial proposal was accepted by a majority of three against one.

    Analog telephony

    All you need to communicate through two phones is a two-wire cable and DC power (a Krona battery, for example). This simple truth is known to communication installers - for decades, each of the representatives of the valiant profession has been hanging in his bag a converted tube from a Soviet disk telephone.

    That is, the very basis of an analog telephone conversation is simple - a carbon microphone is installed in the handset, its resistance changes under the influence of sound waves - current is modulated. At the other end, under the influence of modulated current, a speaker vibrates - a telephone capsule. The first telephone exchanges implemented exactly this scheme - the telephone operator connected the cords of two subscribers.

    Of course, in modern phones, the microphone may well turn out to be piezoelectric, the phones carry out tone rather than pulse dialing, which means that the dial circle does not suit them, and indeed - the phone may not be analog at all.

    In disk phones, the number was dialed by a series of pulses , which in the handset was heard as a series of clicks. Pulse dialing originated during the decade-step automatic telephone exchanges . Pulses from the telephone directly controlled the dialing process at the station. Decade-step automatic telephone exchanges were the first automatic telephone exchanges.

    Today, digital exchanges and IP-PBXs are used. That is, of course, in the dense forests of the immense Homeland, one can find automatic telephone exchanges of a ten-step system, coordinate automatic telephone exchanges and other systems, but they are not installed at new facilities, but they are trying to replace them in cities.

    The PBX in the telephone path at the initial stage played the role of a power source and a switch - between the subscribers of one PBX, the connection established one electric circuit. Of course, everything was reflected in the quality of communication - twists, station devices, interference, thermal currents, etc.


    Digital transmission has gained a road to life on highways - transmission of an analog signal over long distances is futile - interference is summed up and the signal becomes impossible to perceive quality. The digital signal is remarkably regenerated - it is enough to recognize the pulses and generate the same sequence - at the output of the repeater the same signal as it was a dozen kilometers ago.

    It is clear that life is not so smooth - a bit or several changes are possible, which requires redundancy in the coding, although this is not so critical for voice, with a low probability of errors.

    The human ear is capable of perceiving sound with frequencies from 20 hertz to 24-25 kilohertz. In accordance with the theorem of Kotelnikov (Nyquist)digitizing a signal requires a sampling frequency twice the frequency of the signal. In fact, for voice transmission, a much narrower band is sufficient - in telephony, a band from 300 Hz to 3 kHz is adopted. That is, in this band are located the main formants that are most significant for the distinguishability of speech. Telephony uses a sampling rate of 8 kHz. When using 8 bits per step, we get 64 kbit / s. In IP telephony, a codec with these characteristics is called G.711.

    Thus, 64 kbit / s is the standard transmission speed of a single voice channel in digital communication technology. Whatever technology you are faced with - multichannel communication, digital telephony, ISDN, PCM - just such bandwidth is allocated everywhere under one voice channel. All transmission speeds in digital telephony are multiples of 64 kbps and now you know why. You also now understand why an ADSL signal using frequencies beyond the voice spectrum cannot be passed through digital paths, and usually ends at the nearest city telephone exchange - telephony involves transmitting a signal with a frequency of no more than 3 kHz.

    Modems and faxes are also designed with these facts in mind. Therefore, when IP-telephony uses the same codec (G711.64 kbit / s), faxes travel through such channels without problems, if there are no problems with the communication channel.

    Digital exchanges work with such streams. If it is possible to connect traditional analog telephones to a digital telephone exchange, at the input their signal will be encoded into a digital one and it will be processed inside the automatic telephone exchange in the same way as signals from digital telephones.

    Analog alarm

    The simplest alarm system is used in analog telephony - you pick up the phone, and from there you hear a joyful, ringing dial tone. A dial tone is sent to you by the PBX, which learned about your interest on the off-hook => closed circuit. You gladly poke the buttons, and the phone also joyfully, of course, snaps out your clicks with pulses, or sings it with combinations of two PBX frequencies. The PBX, itself or with the goods, quickly understands the essence of your clicking and singing message and, having found your addressee, gives a signal to his device that it is time to ring, what this device will do with all the puppy joy. In the meantime, the called party’s device will twitch in the paraxisms of contentment, you will hear long beeps that will soothingly inform you that the PBX has not forgotten about you. If the device twitched in vain,

    It is easy to see that this type of analog alarm system is by no means designed for automation. Short and long beeps are guaranteed not to be standard - after all, a short / long characteristic is completely perceived by a person. And that means there’s nothing to worry about.

    Guided by such considerations, the creators of telephone systems gave an unforgettable experience to the administrators of VoIP gateways - in the Addpac forums you can read detective stories on recording the busy tone from the PBX and its subsequent analysis by various audio editors. In fact, everything is not so scary with the end.

    But where it really becomes very disappointing - this is dialing to a subscriber of a traditional telephone network. There is such a variable call_timeout in FreeSWITCH (there is a similar in the Dial parameters in Asterisk), which sets the time for trying to get through. If within say 20 seconds no one picked up the phone, then the call will go to voicemail, for example. So, if you set up call forwarding to your mobile device, and if you want to transfer the call to voicemail in case of failure, nothing will work. It is practically impossible for the gateway to determine whether the call is on or someone else has picked up the phone (unless it’s certainly not a GSM gateway, everything is fine with GSM gateways with signaling). It is also difficult to determine by short beeps whether the subscriber is busy, or hangs up after a call.

    Yes, of course, you can make all sorts of detectors for the appearance of voice in the line, but if I do not immediately understand that at the other end someone deigned to reach the handset, then automation, with its percentage of error, is in no way suitable for serious use - it will turn out badly when the caller has already picked up the phone, and the automation is still pondering - “is this a voice, or not a voice? Probably still not a voice, we wait further. ”

    Thus, my attitude to the prospects of analog joints with city exchanges is visible - there are no prospects - analog signaling in the digital age is atavism, why did these crutches, when the PBX knows for sure, picked up the phone or not. The whole question is only that she does not report this in any way, counting on the fact that the apparatus has a person who recognizes a fellow.

    Obviously, the joint must be done either digitally (well, from the options I only saw E1, I need to look for the same R1.5 and PRI, although perhaps there is also an ISDN BRI somewhere), or make a VoIP joint. Now, many providers provide landline phones via VoIP.

    In this series, the fundamentals and the Kotelnikov theorem were mentioned , and Montmorency performed the feat by clamping his teeth with his cable and saved the selector. In the next series, scary stories about codecs, SIP and SDP, as well as the continuation of the series about installer George.

    Original on my personal blog

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