Formant group travel along the vocal tract
Three communications installers. The suffering of George and Harris. The victim of one hundred and seven interferences. Useful recipes. Means against diseases of the vocal tract in installers. Installers agree that they are overworked and that they need rest. A week at sea, away from twists? George offers a trip along the river. Montmorency makes an objection, catches a cell phone on the river. The original proposal was accepted by a majority of three to one.
All that is needed to communicate through two phones is a two-core wire and DC power (the “Krona” battery, for example). Communications installers know this simple truth - for decades, each of the representatives of the glorious profession has a reworked tube from a Soviet dial-up phone dangling in a bag .
')
That is, the very basis of an analog telephone conversation is simple - a carbon microphone is installed in the handset, under the action of sound waves, its resistance changes - current is modulated. At the other end, under the action of the modulated current, the speaker vibrates - a telephone cap. The first telephone exchanges implemented this very scheme - the telephonist connected the cords of two subscribers.
Of course, in modern phones, the microphone may well be piezoelectric, phones perform tone, rather than pulse dialing, which means the dial does not suit them, and in general - the phone may not be analog at all.
In disc phones, the number was dialed by a series of pulses , which was heard in the handset as a series of clicks. Pulse dialing originated during the decade-step PBX . The impulses from the telephone directly controlled the dialing process at the station. Decade-step PBXs were the first automatic telephone exchanges.
Today, digital PBX and IP-PBX are used. That is, of course, in the deaf forests of the immense Motherland, you can also find the ATC of a ten-step system, coordinate PBX and other systems, but they do not install them in new facilities, and in cities they try to replace them.
At the initial stage, the PBX in the telephone path played the role of a power source and a switch — the connection established one electrical circuit between subscribers of one PBX. Of course, everything was reflected in the quality of communication - twisting, station devices, pickups, thermal currents, etc.
The digital transmission received life on highways - the transmission of an analog signal over long distances is unpromising - the interference is summed up and the signal becomes impossible for quality perception. 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 not everything is so smooth in life - a bit or several changes are possible, which requires redundancy in coding, although this is not so critical for voice, with a low probability of errors.
The human ear is able to perceive sound with frequencies from 20 hertz to 24-25 kilohertz. In accordance with the Kotelnikov (Nyquist) theorem, to digitize a signal, a sampling frequency of twice the signal frequency is required. In fact, for voice transmission, a much narrower band is enough - in telephony, the band from 300 Hz to 3 kHz is accepted. That is, in this band, the main formants are located, which are most significant for the intelligibility of speech. In telephony, a sampling rate of 8 kHz is used. When using 8 bits per step, we get 64 kbps. In IP telephony, a codec with such characteristics is called G.711.
Thus, 64 kbps is the standard transmission rate of a single voice channel in digital communication technology. No matter what technology you encounter - multichannel communication, digital telephony, ISDN, PCM - everywhere such a bandwidth is allocated for one voice channel. All digital telephony transmission rates are multiples of 64 kbps and now you know why. Also, you 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 PBX - 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 the same codec is used in IP telephony (G711.64 kbps), faxes travel through such channels without problems if there are no problems with the communication channel.
Digital PBXs work exactly with such flows. If you can connect traditional analog telephones to a digital PBX, their input signal will be encoded in digital and the PBX will be processed in the same way as signals from digital telephones.
The simplest alarm system is used in analog telephony - you pick up the phone, and from there you hear a joyful tone to the dialing number. A dial tone is sent to you by the PBX, which has learned about your interest in the off-hook => closed circuit. You are happy to poke buttons, and the phone also joyfully, of course, snaps off your presses with impulses, or sings it with combinations of two frequencies of ATS. A PBX, by itself or with companions, quickly understands the essence of your clicking-singing message and finding your addressee, sends a signal to his apparatus that it is time to ring out what this apparatus will do with all the puppy joy. In the meantime, the device of the called subscriber will twitch in paraxisms of contentment, you will hear long beeps, which will be soothing to inform you that the PBX has not forgotten about you. If the device was twitching in vain, and no one will pick up the phone, you will receive a series of short beeps.
It is easy to see that this type of analog signaling is not designed for automation. Short and long beeps are guaranteed not standard - because the short / long characteristic is quite perceived by the person. And that means there is nothing to care about.
Guided by such considerations, the creators of the telephone systems presented an unforgettable experience to the administrators of VoIP gateways - in the Addpac forums you can read detective stories about recording the busy tone from the PBX and its subsequent analysis by various audio editors. In fact, everything is not so terrible with the rebound.
But where it really becomes a shame - it is dialing to the subscriber of the traditional telephone network. There is such a variable in FreeSWITCH call_timeout (there is a similar in the Dial parameters in Asterisk), which sets the time for the call attempt. If within 20 seconds we say no one picked up the phone, the call will go to voicemail, for example. So, if you set up call forwarding to your cellular device, and you want to transfer the call to voice mail in case of failure, nothing will work. It’s virtually impossible for the gateway to determine if the call is on, or if someone has already picked up the phone (if this is not a GSM gateway, of course, everything is good in GSM gateways with signaling). It is also difficult to determine whether the subscriber is busy with a short beep, or hang up after a call.
Yes, of course, you can tinker with all sorts of detectors of voice in the line, but if I do not immediately understand that at the other end someone deigned to reach the tube, then automation, with its percentage of error, is in no way suitable for serious use - it will turn out bad the callee has already picked up the phone, and the automation is still thinking - “is this a voice, or is it not a voice? Probably still not a voice, we wait further. ”
Thus, my attitude towards the prospects of analog joints with urban exchanges is clear - there are no prospects - analog signaling in the digital age is an atavism, why these crutches, when the automatic telephone exchange knows for sure whether they picked up the phone at the end or not. The only question is that it does not inform about it in any way, hoping that the apparatus has a person who recognizes a colleague.
Obviously, the joint should be done either digitally (well, I only saw E1 from the options, and that R1.5 and PRI still need to be searched, although, perhaps, somewhere else there is also ISDN BRI), or to make a VoIP interface. Now many providers provide landline phones via VoIP.
In this series, the fundamentals were stated and the Kotelnikov theorem was mentioned, and Montmorency accomplished the feat, holding the cable with his teeth, 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.
Analog telephony
All that is needed to communicate through two phones is a two-core wire and DC power (the “Krona” battery, for example). Communications installers know this simple truth - for decades, each of the representatives of the glorious profession has a reworked tube from a Soviet dial-up phone dangling in a bag .
')
That is, the very basis of an analog telephone conversation is simple - a carbon microphone is installed in the handset, under the action of sound waves, its resistance changes - current is modulated. At the other end, under the action of the modulated current, the speaker vibrates - a telephone cap. The first telephone exchanges implemented this very scheme - the telephonist connected the cords of two subscribers.
Of course, in modern phones, the microphone may well be piezoelectric, phones perform tone, rather than pulse dialing, which means the dial does not suit them, and in general - the phone may not be analog at all.
In disc phones, the number was dialed by a series of pulses , which was heard in the handset as a series of clicks. Pulse dialing originated during the decade-step PBX . The impulses from the telephone directly controlled the dialing process at the station. Decade-step PBXs were the first automatic telephone exchanges.
Today, digital PBX and IP-PBX are used. That is, of course, in the deaf forests of the immense Motherland, you can also find the ATC of a ten-step system, coordinate PBX and other systems, but they do not install them in new facilities, and in cities they try to replace them.
At the initial stage, the PBX in the telephone path played the role of a power source and a switch — the connection established one electrical circuit between subscribers of one PBX. Of course, everything was reflected in the quality of communication - twisting, station devices, pickups, thermal currents, etc.
Numeral
The digital transmission received life on highways - the transmission of an analog signal over long distances is unpromising - the interference is summed up and the signal becomes impossible for quality perception. 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 not everything is so smooth in life - a bit or several changes are possible, which requires redundancy in coding, although this is not so critical for voice, with a low probability of errors.
The human ear is able to perceive sound with frequencies from 20 hertz to 24-25 kilohertz. In accordance with the Kotelnikov (Nyquist) theorem, to digitize a signal, a sampling frequency of twice the signal frequency is required. In fact, for voice transmission, a much narrower band is enough - in telephony, the band from 300 Hz to 3 kHz is accepted. That is, in this band, the main formants are located, which are most significant for the intelligibility of speech. In telephony, a sampling rate of 8 kHz is used. When using 8 bits per step, we get 64 kbps. In IP telephony, a codec with such characteristics is called G.711.
Thus, 64 kbps is the standard transmission rate of a single voice channel in digital communication technology. No matter what technology you encounter - multichannel communication, digital telephony, ISDN, PCM - everywhere such a bandwidth is allocated for one voice channel. All digital telephony transmission rates are multiples of 64 kbps and now you know why. Also, you 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 PBX - 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 the same codec is used in IP telephony (G711.64 kbps), faxes travel through such channels without problems if there are no problems with the communication channel.
Digital PBXs work exactly with such flows. If you can connect traditional analog telephones to a digital PBX, their input signal will be encoded in digital and the PBX will be processed 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 tone to the dialing number. A dial tone is sent to you by the PBX, which has learned about your interest in the off-hook => closed circuit. You are happy to poke buttons, and the phone also joyfully, of course, snaps off your presses with impulses, or sings it with combinations of two frequencies of ATS. A PBX, by itself or with companions, quickly understands the essence of your clicking-singing message and finding your addressee, sends a signal to his apparatus that it is time to ring out what this apparatus will do with all the puppy joy. In the meantime, the device of the called subscriber will twitch in paraxisms of contentment, you will hear long beeps, which will be soothing to inform you that the PBX has not forgotten about you. If the device was twitching in vain, and no one will pick up the phone, you will receive a series of short beeps.
It is easy to see that this type of analog signaling is not designed for automation. Short and long beeps are guaranteed not standard - because the short / long characteristic is quite perceived by the person. And that means there is nothing to care about.
Guided by such considerations, the creators of the telephone systems presented an unforgettable experience to the administrators of VoIP gateways - in the Addpac forums you can read detective stories about recording the busy tone from the PBX and its subsequent analysis by various audio editors. In fact, everything is not so terrible with the rebound.
But where it really becomes a shame - it is dialing to the subscriber of the traditional telephone network. There is such a variable in FreeSWITCH call_timeout (there is a similar in the Dial parameters in Asterisk), which sets the time for the call attempt. If within 20 seconds we say no one picked up the phone, the call will go to voicemail, for example. So, if you set up call forwarding to your cellular device, and you want to transfer the call to voice mail in case of failure, nothing will work. It’s virtually impossible for the gateway to determine if the call is on, or if someone has already picked up the phone (if this is not a GSM gateway, of course, everything is good in GSM gateways with signaling). It is also difficult to determine whether the subscriber is busy with a short beep, or hang up after a call.
Yes, of course, you can tinker with all sorts of detectors of voice in the line, but if I do not immediately understand that at the other end someone deigned to reach the tube, then automation, with its percentage of error, is in no way suitable for serious use - it will turn out bad the callee has already picked up the phone, and the automation is still thinking - “is this a voice, or is it not a voice? Probably still not a voice, we wait further. ”
Thus, my attitude towards the prospects of analog joints with urban exchanges is clear - there are no prospects - analog signaling in the digital age is an atavism, why these crutches, when the automatic telephone exchange knows for sure whether they picked up the phone at the end or not. The only question is that it does not inform about it in any way, hoping that the apparatus has a person who recognizes a colleague.
Obviously, the joint should be done either digitally (well, I only saw E1 from the options, and that R1.5 and PRI still need to be searched, although, perhaps, somewhere else there is also ISDN BRI), or to make a VoIP interface. Now many providers provide landline phones via VoIP.
In this series, the fundamentals were stated and the Kotelnikov theorem was mentioned, and Montmorency accomplished the feat, holding the cable with his teeth, 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.
Source: https://habr.com/ru/post/51129/
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