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In this article, I will try to consider the basic principles of the telephone, describe the most commonly used protocols, explain the methods of speech encoding and decoding, and analyze some typical problems. -Telephone refers to voice communication, which is carried out through a data transmission network, especially communication through a network (-protocol). Today, telephones are gradually replacing traditional telephone networks due to easy deployment, low call costs, easy configuration, high communication quality and high connection security. In this presentation, we will follow the principles of the reference model (basic reference model for open systems interconnection) and discuss the topic from bottom to top, starting with the physical layer and data link layer and ending with the data layer. When making a call, the voice signal is converted into compressed data packets (this process will be discussed in more detail in the chapters "Pulse Code Modulation" and "Codec"). Next, the packet data is forwarded through the packet switching network, especially the network. When the data packets arrive at the receiver, they are decoded into the original voice signal. These procedures are possible due to a large number of auxiliary protocols, some of which will be discussed below. A language that allows two subscribers to understand each other and provides high-quality data transmission between two points.
In traditional telephones, the connection is established using a telephone exchange, and only the purpose of dialogue is pursued. Here, the voice signal is transmitted on the telephone line through a dedicated connection. In the case of a telephone, the compressed data packet will enter the global or local network at a specific address and be transmitted according to that address. In this case, addressing and all its inherent functions (such as routing) have been used.
It turns out that-it is a cheaper solution for both operators and subscribers.
Traditional telephone networks provide redundant performance, while telephones use voice data packet compression technology to fully utilize the capacity of the telephone line. Generally, everyone can access the global network, which makes it possible to reduce the cost of connection or eliminate it completely.
Calls in the local network can use internal servers and can be made without external.
Telephony servers are constantly improving, and their algorithms are becoming more resistant to delays or other problems in the network.
In a private network, the owner has complete control over the situation and can change parameters such as bandwidth, the number of subscribers on a line, and the amount of delay caused by this. Packet switching networks are developing, and new protocols and technologies are introduced every year to improve communication quality (for example, bandwidth reservation protocols).
-Very elegantly solved the problem of busy lines, because you can execute forwarding or switch to standby mode through several commands in the above configuration file. 
At the physical layer, the bit stream is transmitted on the physical medium through the corresponding interface. The telephone is almost entirely dependent on the existing network infrastructure. As the medium of information transmission, Category 5 twisted pair (5), single-mode or multi-mode fiber or coaxial cable is usually used. Therefore, the convergence principle of telecommunications networks can be fully realized.
Its essence lies in being able to supply power to the device through standard twisted pair cables. Most modern phones, especially the 7901 series, have support. According to the 2011 standard, the device can provide up to 26.5 watts of power. When supplying power, only two twisted-pair 111- cables are used, but some manufacturers use all four twisted-pair cables with a maximum power of 52 watts. It should be noted that this technology does not require modification of the existing cable system, including Category 5 cables. In order to determine whether the connected device is powered (powered device), a voltage of 2.8-11 is applied to the cable to calculate the resistance of the connected device. If this resistance is in the range of 20-27.5, the process enters the next stage. If not, repeat the inspection at ≥2 intervals. Next, perform a search for the power range of the powered device by applying a higher voltage and measuring the current in the line. After that, a voltage of 49 is applied to the line-supply voltage. Overloads are also continuously monitored.
A device that provides the connection of multiple nodes of a computer network and the distribution of frames between hosts based on physical addressing.
This technology allows you to create logical network topologies without considering their physical properties. This can be achieved by marking traffic, which is described in detail in the 803.1 standard. It is widely used to isolate voice traffic and other data generated by the phone.
Reduce the possibility of intercepting and analyzing voice data packets. Allows you to set a higher priority for voice packets, thereby improving communication quality.
At the network level, routing is selected separately, and the main equipment at the network level is the router. This is where it is determined how the data reaches the recipient through a specific address.
Connect a traditional analog phone to the network. Usually, they also have built-in routers that allow them to track traffic, authorize users, automatically assign addresses and manage bandwidth. It should be supplemented with other tools (such as queuing protocols) (so that voice data does not compete with regular data).
Usually, for this, routers use low-latency queues or class-based weighted fair queues.
In addition, a priority coding scheme is required to treat voice data as essential for transmission. The point is that they cannot provide reliable data transfer. This is a better choice than delivery control because the phone is highly dependent on transmission delay, but is less sensitive to packet loss. Provide non-guaranteed delivery, that is, no confirmation is required when sending and receiving data. Similarly, when sending information, there is no need to establish a logical connection between the modules (source and target). It is usually regarded as a transport layer protocol and usually runs on top of. Using, you can realize traffic type identification, use time stamp, transmission control and data packet sequence number. 
It contains the fact that it assigns a timestamp to each outgoing packet and processes it on the receiving side. This can receive data in the correct order, reduce the impact of uneven packet transmission time, and restore synchronization between audio and video data.
Such combinations are allowed because the processes that occur at these levels are closely related to each other, and it would be more logical to describe them without considering their division into sub-levels.
This standard contains descriptions of equipment, network services and terminal equipment intended for audio and video communications in packet-switched networks. For any device of the .324 standard, it needs to support voice information exchange.
These recommendations do not define the physical transmission medium, transmission protocol and network interface. This means that devices that support the .324 standard can work in any packet-switched network that exists today.
The two bit rate speech encoders used for multimedia communication have bit rates of 5.3 and 6.3 respectively.
Use linear prediction and low-delay excitation signal for encoding.
Algebraic coding linear prediction of excitation signal with conjugate structure.
Video encoding for low bit rate transmission.
Provides security and encryption for multimedia terminals in the .324 network. .324 Supplementary general functions of service management.
Transfer the connection to the third party's phone number.
Incoming call notification when in a call.
A signaling protocol designed to organize, change, and terminate communication sessions. It has nothing to do with the transmission technology, but it is preferred when establishing a connection. For the transmission of voice and video information itself, it is recommended to use, but it does not rule out the possibility of using other protocols. Send information about the user's location to the server, which can broadcast it to the location server.
Audio codecs are programs or algorithms that compress or decompress digital audio data to reduce bandwidth requirements. In the phone, by far, the most common conversion is through the .730 codec and .712 compression based on the law and μ law.
It is a codec that compresses the original signal in case of data loss. The main idea inherent in .730 is not to transmit the digitized signal itself, but to transmit enough parameters (spectral characteristics, zero crossing times) for subsequent synthesis at the receiving end. At the same time, all the basic characteristics of speech, such as amplitude and timbre, are retained. The channel bandwidth designed for this codec is 8. The processed .730 has a frame length of 11 and a sampling frequency of 8. For each of these frames, the parameters of the mathematical model are determined and then transmitted to the channel in code form.
It places high demands on processor resources. 
The speech codec does not perform any compression other than compression. This is a method to reduce the influence of channels with limited dynamic range. This method is based on the principle of reducing the number of signal quantization levels in the high-volume area while maintaining sound quality. Two companding schemes widely used in telephones are and. The signal in this codec is provided by 65 streams. The sampling rate is 8001 frames at 8 bits per second. The voice quality is subjectively better than using the .730 codec. Lossy compression algorithm for audio data. Mainly used in Japan and North America. The continuous function is transmitted in the form of a series of continuous pulses. In order to receive the modulated signal at the input of the communication channel, the instantaneous value of the carrier signal is measured within a certain period. In this case, the number of digitized values per second (otherwise the sampling rate) must be greater than or equal to twice the maximum frequency in the analog signal spectrum.
In addition, the obtained value is rounded to one of the previously accepted levels. Please note that the number of levels must be a multiple of a power of 2. According to the defined number of levels, the signal will be encoded with a certain number of bits.
The figure shows the use of four-bit encoding (that is, all intermediate values of the analog signal will be rounded to one of the predefined 17 levels). For example, if time is equal to zero, the signal will be represented in the following way: 74.
During demodulation, the sequence of zeros Beattock becharof roadmik and ones is converted into pulses by the demodulator, and its quantization level is equal to that of the modulator. Afterwards, the signal is reconstructed from these pulses, and then the smoothing filter finally eliminates the inaccuracy. In modern phones, the number of quantization levels must be greater than or equal to 111, that is, the minimum number of bits that can be encoded on a signal is 7.
By default, high-quality services for delay-sensitive traffic are not provided. When using the protocol, the reliable delivery of information can be guaranteed, but unpredictable delays may occur in the delivery of information. It tends to minimize the delay, but cannot guarantee correct packet delivery.
At the same time, the quality factor of voice traffic depends to a large extent on the transmission quality, and in a network where there is no implementation mechanism to ensure proper quality, the implementation of telephone may not meet the requirements of users. The main indicators of service quality are network throughput and transmission delay. In this case, the delay is defined as the elapsed time from sending the data packet to receiving the data packet. There are also features such as network availability and reliability (evaluated by monitoring service levels for a long time or by utilization). When one of the communication channels is overloaded, it allows the use of alternate routes for transmission. Reserve communication channel resources during the connection.
Provides functions to mark packages according to their importance and perform label-based maintenance. As mentioned earlier, voice traffic is very sensitive to transmission delay. The maximum delay time should not exceed 401 milliseconds (including the duration of terminal station information processing).
The delay of information encoding in the voice gateway or terminal equipment. Reduce by improving speech processing and conversion algorithms.
By improving the network infrastructure, especially by reducing the number of routers and using high-speed links to reduce.
Jitter, or in other words, random packet propagation delay.
The most common way to deal with jitter is a jitter buffer that stores a specified number of packets.
Usually, the length of the buffer is dynamically adjusted during the entire life cycle of the connection. A heuristic algorithm is used to select the optimal length. 
To compensate for the uneven packet arrival rate, temporary packet storage or so-called jitter buffers are created on the receiving side. Its task is to collect incoming data packets in the correct order according to the timestamp, and send them to the codec at the correct interval and in the correct order. The device performs calculations during operation, or mandatory settings in the settings. On the one hand, it cannot be too large so as not to increase the transmission delay. On the other hand, when the network latency changes, a smaller buffer size will cause packet loss.
This is one of the main conflicts between providers and phone users. From the perspective of the provider, all packages have been delivered to the subscriber, that is, there is no loss. From the device's point of view, the time difference between the arrival of data packets is much larger than the jitter buffer. Therefore, there is actually a loss. In fact, a loss of more than 1% will cause some discomfort. When it reaches 2%, dialogue becomes difficult. Over 4%, dialogue is almost impossible.
Let us give an example to calculate the expected size of the random propagation delay of the fifth packet based on the first two examples.
On average, the random propagation time delay of a packet in the current example is 11 milliseconds (more precisely, you can use the formula above to calculate). Then, in order not to lose any data packets, the size of the jitter buffer should be equal to 11.
To determine the required jitter buffer size (in megabytes), multiply the resulting value by 111-average network bandwidth: 11•11^ -3•111 = 139. The size of the jitter buffer must be greater than the fluctuation of the network transmission time. For example, if the transmission time range of 11 packets is 5 to 11 milliseconds, the buffer must be at least 8 milliseconds so that no packets are lost. If the buffer is larger (for example, 13) is better, then the mechanism of re-requesting lost packets will work.
-Telephone and traditional public telephone network. Dynamically develop open source software that can be installed regardless of license. This makes the software attractive to small and medium enterprises. The number of subscribers in the network can reach 2001 and is only limited by server capacity.
All necessary functions are either already implemented or can be added independently without spending a lot of time and money. The principle facilitates this: one task-one software module. Compared with the solutions of Cisco or other vendors, the cost of deployment is also very attractive. In fact, just buying a phone and a server that can provide the required load on the network can reduce all costs. The program itself is completely free.
Instead, it is suitable for large networks with 30,000 subscribers. This sophisticated hardware and software provide reliable operation and allow you to configure many parameters, such as call forwarding or voice menus. There is also a "lightweight" fast version, which is more used in small offices.
The first thing to note is the well-known technical support of Cisco. With an appropriate level of service contract, everything from installation problems to equipment failures can be solved almost immediately. Therefore, it is suitable for companies that are prepared to pay a lot of money but can get the highest quality of service.
It may be a good choice for medium-sized telephone networks. The number of subscribers is not only limited by server capacity, but also by the number of purchased licenses. You need to license almost everything-expansion cards, applications used, etc., which may cause some inconvenience. There are many programs to configure, but the most popular and easiest to use is. You can also use the control console. It has been a recognized leader in the telephone equipment market since it merged with another well-known manufacturer, Nortel. Now, this is very useful to me because I am writing a diploma on telephony and I am testing and comparing different solutions for building small networks.
In terms of management, I encountered a complete misunderstanding. As a result, we purchased a simulation. Maybe this sentence is useful for your diploma.
Therefore, one will not interfere with the other.
Does it have any advantages over the asterisk? And how to be friends with providers (e.g. and-). Maybe in the next article. This will be useful.
To be honest, I have never worked with, so I have nothing to say. As for the asterisk-I think he is a friend of everything, although there are certain problems in trying to make him a cis friend.
As a result, in the past three years, I encountered such a problem instead of paying 81 fees. Bought a gateway of 31. As far as analog Panasonic is concerned-cool radio station, trustworthy. They are also reliable for digital communications, but compatibility and customization flexibility are of course weak.
In my opinion, this article is completely useless. It is still impossible to understand its working principle and function. It would be better to apply pros/cons.
The essence is that the sent data packets may arrive at the receiver in the wrong order for many reasons.
Under ideal conditions, no matter what, packets sent at 21 intervals will be received by the receiver at the same 21 intervals, and they can be decoded and sounded immediately. In fact, the first packet will arrive in 51 milliseconds, the next 41 milliseconds, the next 151 milliseconds, and so on. In this case, you will not be able to play the sound immediately, otherwise the problem is that the packets arriving within 151 will be dropped. We need a buffer into which data packets can be typed, thereby increasing the latency of the session. Probably not, but we often encounter package reordering. In a real sense, that is to say, in a dozen packages, at least two will be mixed together? The tolerance for this is poor-it is usually equated with a lost packet. And if occasionally during rerouting, when the first data packet passes through one link, and the +1 and subsequent data packets pass through another link, this is normal. Collect package dumps, start the application, and kick it every hour. Or it may not be repairable, but everyone is used to it.
In the process, kicked to the top of the company-it did not help either. The company’s engineers swear that there is no parallel path anywhere.
The company’s engineers swear that there is no parallel path anywhere.
Multiple interfaces on different parts of the path, at most once. We have performed more than one operation, for example, in the case of falling somewhere, we don't know where the channel is. After that, the problem is rarely resolved within a day. In this case, we call the service manager and department head to the office to have a dialogue with them on the subject, and the problem will be automatically eliminated. In today's Internet, this is not an infinite value. It happened.
And the duplex mismatch somewhere in the operator's network (they spent a few days looking for the reason for the disconnection under low load).
It rarely happens, or we rarely notice it. Everything is normal, sometimes it only slows down. To connect analog users in the network and connect ours to the provider's analog line). Can analog fax and modem work in this configuration? Which protocols (.31, .39) must the gateway support for this? What is the maximum speed available? Thank you in advance. However, this does not mean that everything will work 101%. As it happens, the different implementations of .39, starting and returning to 712, make fax transmission impossible. In addition, if there is on the way, it is very likely that 96% of faxes will not be transmitted via .39. Only the voice works hard. It supports fax itself without switching to .39, so if 712 is used in the office network and the gateway supports it, then in theory (so far, practice has proven), fax should work. If the priority of the network is 731, the gateway should be able to use .39, but various strange problems will occur during the exchange, so the operability cannot be guaranteed 101%.
The diploma will also be a similar subject, in addition to the practical part, it also requires a lot of mathematical knowledge, I want to consider a few agreements. Can you tell me about this source of quality? Russian/English books are best. Be prepared to open the ticket and wait at least one hour for the engineer to contact you (even if the priority is high). Then prepare your ticket, depending on the complexity, from one day to several months. Moreover, if there is no subject on the ticket, then they will probably not be able to resolve it. For Yemen, I would not say that they are traditionally wooden, but very capable people sit in ours. It is very simple, easy to use, very stable (I haven't seen any), and it scales perfectly. This is per cluster, not per server.
-Change some parameters of hundreds of phones, and the value of this parameter is different for all 5 phones? Personally, two weeks ago, I used to submit two bugs. Now, another mistake with the programmer is in trouble. I updated it this weekend (rolling). Therefore, during the restart process, it fell into a coma for half an hour, and then she said that the service was not started. Restart-everything is like clockwork. You have no time to buy, and the developer fixed the error.
Better yet, think about the architecture, because it allows this, so there is no need to do this. I do not need
Personally, two weeks ago, I used to submit two bugs. It is necessary to transfer the product to the new major version immediately.
And most of the errors are fixed without major updates. Vaughan is still playing under 7 years old.
I once hung up the phone for 1.5 hours in case of a network failure, and then waited to find an engineer. 
That is, it turns out that in order to update a parameter, you need-export the phones, delete them, import the phones (and pray that the importer will not be damaged). At this time, the user is sitting where there is no phone. I don't do this, I prefer to configure the phone through. But this is to write a program every time. It is not easy to configure. No-this means that there is not enough operating experience. And check how many tickets the mature version has in.
There is an infrastructure for thousands of phones and hundreds of gateways. But I was thinking-I don't know why this is happening.
I tried to catch her a few times and I was already very happy. I posted the trail (the only disgusting trace in) to immediately fix everything on the case and found my joints. Also, it is not easy to filter it out in order to start changing the two groups.
After all, did someone activate all of this from it and then fix it? Okay, this is a temporary error-I haven't seen 7- yet, but I can almost be sure that the site is being applied (or not at all) correct on the homepage. There is no design at all. This is the case for both version 6 and version 8. I know the error is a cosmetic error, but it surprised me-it is so obvious and easy to fix. but not. , And then apply the desired to the group with one click. And I can hardly imagine anything more convenient. You still need to make a list of numbers.
So what can you say.
Ohm, and many equally important things related to the phone, but about the model and advertising quotations, and the specific connection with the phone, you need to read it separately. Article about everything-nothing in the end. Generally, for those who are interested, I suggest to increase with the minimum configuration, install the softphone, and simply observe how the connection is established and what happens in the process will be very helpful. For people who don't understand the model and stack, I usually don't recommend you to use network technologies and protocols.
So far, it is the best in terms of the transmission rate of voice information. The network is good, but each node will introduce a significant delay in signal propagation. In fact, the network is only suitable for transmitting voice services when the gigabit channel is introduced without causing discomfort to users. Until now, real telecom operators still comply with the rule: it is acceptable in short distances (regions with villages), and everything further (long distance/international communication) is still a network with 1. However, there is something to understand. Traditional operators must meet strict requirements for communication quality. The network will not always be able to resolve there, especially when you consider that in addition to the volume of voice traffic, (unpredictable in terms of volume) also flows through the network.
It has been implemented nationwide, and the traditional network has not actually developed.
So far, it is the best in terms of the transmission rate of voice information.
Now everyone is actively moving the intercity to. Cheaper and more convenient. I did not even talk about the fact that even in my office in Moscow, the central center can communicate with Vladivostok and other remote Moscow loops through the tunnel through ordinary communication channels, and the communication characteristics will be in Higher level. As far as the speed of voice information transmission is concerned, it is the best. When talking about speed, I imagine gigabytes instead of milliseconds. There is no particular problem. Yes, some operators like to multiplex 1 stream, but everyone is trying to get rid of this madness.
Internet traffic on the network is also difficult to predict. When using this channel, no voice packets will be dropped or delayed in the queue. This is not a problem for large operators such as those on their own networks.
It's just that the ratio of voice traffic to data traffic is not suitable for the first one, so in order not to build a different transmission network, they will switch to. The reliability and routing redundancy are also higher and faster.
This further hinders the need for a separate infrastructure. A properly configured packet network is close to these values. Converges within about 51. This can already be said.
The operator's is too high, especially if you send it to everyone (for example, to activate at 10010 per day)? But what does messaging have to do with the telephone? Maybe I just don't know anything about it, but that doesn't help sending mailing lists to 1001 phones cheaply. Or I don't know-tell me. Can replace any content of and. It is just the simplest protocol, most similar to the protocol used by operators for transmission, and requires the least number of intermediate layers (but still bad).
So, at least in many files. This also means the division into random and stable delays, which is not very obvious and is not entirely correct.
When setting the size of the jitter buffer, the network's response to the accident should be considered. For example, if the convergence time is 51, a larger buffer size should be set. You also need to look at how the device works and when to re-request lost packets. For example, by default, packets on the radio are very large-up to 0.5, which is too much for.
The delays are quite comfortable, but they take all the delays into account, and the size of the jitter buffer has been added to the average network delay. In fact, I had no problems when increasing the jitter buffer to 41-51.
In competition, no matter how many subscribers you have-you can score 100,000 without any problem, what matters is what kind of traffic you want to get-roughly speaking, how many calls are there at the same time, and depending on capacity and adjustments, the server can accommodate one One hundred to one thousand.
Each phone giant can at least provide its own phones, because their hardware is the same, so the main content is the distribution of load on the cluster. Inside a Google data center