"Pick To: From the perspective of application, this paper puts forward a new software and hardware design scheme of LAN IP phone, that is, taking at75c220 development board as the hardware platform, using H.323 protocol stack to develop LAN IP phone terminal under mclinux operating system, and lists some efficient processing methods for the key problems in the implementation.
preface
Because the development and update of computer network technology is faster than PBX and the price of equipment decreases greatly, it has become the goal of enterprises to establish an IP telephone network based on Internet (or intranet) to replace the traditional telephone equipment system.
LAN IP phone
LAN IP phone uses unified network communication equipment and wiring to transmit voice and data. In the traditional PBX (private branch exchange) system, the voice call enters the office through a series of standard voice lines connected with the office PBX, that is, it receives and dredges the voice traffic on the standard telephone wiring through a special equipment. However, in LAN telephone voice network, voice calls are received and channeled through IP based PBX, which is connected to the existing data network. IPPBX can be an independent solution or can be decomposed into several decentralized devices. The specific functional block diagram is shown in Figure 1.
The main advantages of this IP phone are:
? It can easily move and add terminal equipment;
? Support multimedia terminal;
? It is conducive to the development of computer telephone integration technology;
Figure 1 function module diagram of LAN IP phone
Figure 2 structural block diagram of at75c220
Figure 3 IP phone structure diagram
hardware platform
Generally, the basic design requirements of hardware are: high density, low power consumption and low cost. Many performances of VoIP correspond to specific application requirements, such as: (1) system segmentation, including packet collection and routing（ 2) Define the software performance of product functions（ 3) Meet the network management capability of high availability.
LAN telephone products work in the pure IP mode, so the telephone should be compatible in network management after accessing the WAN. The VoIP phone of this scheme is a terminal product with gateway down. This device can be directly connected to Ethernet without gateway, and can be used after configuring relevant address information. Its development is based on ATMEL's at75c220 development board. At75c220 is a high performance processor chip specially designed for various Internet applications. Its core component, ARM7TDMI microprocessor, has a running capacity of 40mips. The oakdspcore coprocessor (voice processing chip in Figure 2) specially used for voice processing has a running capacity of 60mips. The efficient dual port mailbox mechanism is adopted for the information interaction between the two. This determines that at75c220 can integrate the control function of general CPU and the information processing function of special CPU, which improves the integration of the system. At the same time, the operation instructions of ARM processor are relatively simple and reduces the complexity of development.
In addition to the standard serial interface for connecting computers and RJ11 interface for connecting telephone handles, at75c220 development board also has two standard RJ45 Ethernet interfaces for connecting gatekeepers. The ARM7TDMI core processor controls the operation of the whole VOIP phone and provides a general I / O interface for connecting dial-up keyboard, LCD and ringing circuit. The structural block diagram of at75c220 is shown in Figure 2.
The at75c220 development board is equipped with flash memory for storing application programs. Using the super terminal program, the computer can interact with at75c220 chip through the serial interface on the development board, so as to realize the application programming of flash and other software management functions. The PC developing at75c220 application software must start and run SIAP mclinux operating system. After the application program is developed, download the IMG file containing the application program and operating system kernel to flash on the 220 development board through the tools provided by 220 software set, and finally complete the system development.
The structure of VoIP phone is shown in Figure 3. The user interface part is no different from the Mandarin machine. The display is used to display relevant information, such as the number dialed when initiating a call, caller ID, etc. In addition, VoIP telephone can be connected with personal digital assistant (PDA) equipment through serial port to realize software upgrading, automatic dialing and other functions. The function of voice interface is to realize the mutual conversion between voice analog signal and standard 64kbit / s PCM signal. The network interface is responsible for the transmission and reception of voice packets and the exchange of various call signaling. The VoIP telephone is connected with the gatekeeper on the LAN through the RJ45 plug.
Figure 4 H. 323 call signaling process
Software implementation
The implementation relies on network protocol stack and real-time operating system (RTOS). Most application systems require RTOS to handle multiple processes and calls at the same time. RTOS adopted should have the following characteristics to meet the complexity of communication protocol: small system kernel; Short interrupt processing time; Long continuous operation time; The processing power of multiple millisecond or microsecond timers. Arm mclinux is an excellent embedded real-time operating system. It provides a variety of functions for real-time system development, debugging and operation, such as multitasking mechanism, kernel tailoring, network function, real memory management strategy and so on. At the same time, the Linux kernel source code is completely open, which is undoubtedly very beneficial to reduce the development cost and improve the reuse degree of software development.
For the network protocol stack, this scheme adopts the standard H.323 protocol stack to interconnect with the public network. From the perspective of TCP / IP layering, H.323 is an application layer protocol family. It contains protocols suitable for various media communication and signaling control. The foundation is TCP or UDP protocol. According to the actual use requirements, the protocols discussed in this paper include G.723.1 and G.711 in speech coding and H.225 and H.245 in call control signaling. Under the normal design capacity of the CPU, all processes of the system will be blocked in their respective message queues. Only the lowest idle process is running, and the total number of messages in the message queue is at a relatively low level. Adding processes may increase the shared data area and internal messages, and introduce complex management mechanisms accordingly.
Based on the above software requirements, the VoIP phone shall be able to communicate with the gateway and gatekeeper conforming to the H.323 standard, and realize the voice communication function between the VoIP phone and the VoIP phone and the Mandarin phone carried by the gateway. In addition, VOIP phones should also have certain recording and playback capabilities.
For IP phones, the main software realizes the communication between the phone and the gatekeeper and the interworking between the phones. The call flow is shown in Figure 4.
The channel implementation mechanism of the above process is TCP or UDP. After the IP phone successfully logs in to the gatekeeper, its connection process is completely similar to the connection between Mandarin phones. After getting off the phone, it sends a "call request" (UDP) to the gatekeeper. After receiving it, the gatekeeper sends "whether to run caller dialing" (TCP). If it is not allowed, it will prompt to hang up; If allowed, the caller starts dialing after receiving the dialing tone, sends the "dialing information" message (TCP) to the gatekeeper, and the gatekeeper sends the "called idle information" message (TCP) to the caller, and then sends back the ring tone and ringing tone to the caller and the called respectively. After waiting for the called to pick up, the called sends the "called off hook" message (UDP) to the gatekeeper, the gatekeeper sends the "stop ringing" (TCP) to the called, and the gatekeeper sends the "called off hook" message (TCP) to the caller to enter the call (UDP) state. The voice processing involves 220 voice processing modules.
The message sent by the gateway to the gatekeeper: whether it is the login message sent by the high-level client to the gatekeeper or the call processing message sent by the IP phone or switch to the gatekeeper, it will be placed in the message queue named mqrecvbottom first, and then sent to the following three queues according to the transmission method adopted by the message: TCP transmission of signaling_ mqSendTCP; UDP transmission of signaling_ mqSendUDP; Voice transmission_ Mqsenddata (can be forwarded by gatekeeper or directly sent between two gateways), and then sent out through socket.
Solution of several key problems
Pick up and hang up detection and dial number reception
At75c220 provides several user-defined general I / O interfaces. Only through certain hardware connection and software programming, the detection of off hook status and the reception of the number dialed by the user can be realized.
Incoming call ring
There are special bits in the internal register of at75c220 to indicate whether there is an incoming call. The application software detects the bit regularly. Once an incoming call is found, it immediately sends a ringing indication signal through a general I / O port determined by programming in advance, which can be used as the enabling signal of the ringing circuit.
Voice processing module
The functions of voice processing are jointly completed by oakdspcore controlled by ARM7TDMI.
During IP call, compress the 64kbit / s PCM digital voice signal sent by the analog front-end circuit and send it to the network interface module; At the same time, decompress the compressed voice signal sent by the network interface module to form a 64kbit / s PCM digital voice signal and send it to the analog front end.
Through the flexible programming of oakdspcore, VOIP phone can easily realize the recording and playback function of voice message.
When the user dials from the dial, start the DTMF signal generator function of oakdspcore by programming to generate the corresponding dual tone multi frequency signal in the user's earpiece. If necessary, the DTMF signal can also be transmitted after compression coding like a voice signal.
G.723.1 algorithm is used for voice compression and decompression. In order to perform effective speech compression, many important factors must be considered. First, when all channels are working, you must ensure that there is no performance degradation. Packets must be configurable to ensure maximum flexibility. In addition, the G.723.1 algorithm used in this subject adopts voice activation detection technology. VAD technology is the basis of adaptive gain control, which can further realize bandwidth compression function and can be used together with adaptive noise generator. VAD technology enables the transmitter to detect the gap of local voice, and no longer send a complete voice frame during this period, but replace it with a static insertion description frame with less bits. This frame only contains the parameters required to generate noise at the decoder input, so that the receiver can generate appropriate background noise, so as to make the call effect close to the real conditions and further reduce the coding rate. The determination of VAD threshold is the key factor to accurately judge sound / silence. For example, continuous speech for a long time will increase the estimated value of background noise and the corresponding threshold, so that the subsequent low amplitude speech is not detected. One solution is to change the cut-off frequency of the low-pass filter when voice is detected, that is, different methods are used to estimate the noise energy when there is sound / silence.
Voice quality
Network delay and jitter are the key factors affecting speech quality. Packaging is also an important factor affecting delay. The real-time performance and efficiency of packaging are a pair of contradictions. How much information is gathered and then packaged is closely related to bandwidth, so an appropriate threshold must be found“ Jitter is a unique phenomenon of packet switching. The method to eliminate jitter is that the receiver uses "jitter buffer" to make up for the unreliability of the packet network. The buffer can be a dynamic queue. The receiver determines the network traffic status according to the RTP timestamp, so as to change the size of the buffer in time. In the specific implementation, a ring queue pointer table can be established to manage the occupied cache. If you want a simple implementation, you can set a fixed number of arrays, and then identify several flag bits for management and control. In addition, echo cancellation is also one of the important aspects. A good echo canceller should have short convergence time and small residual echo, reliably detect stress, and be able to process background noise and narrowband signals. In this subject, the G.165 algorithm in DSP is used to eliminate voice echo to the greatest extent.
epilogue
As a new communication terminal product, LAN IP phone has a lot of development space. At present, there are many sample forms, but the general trend is: simple hardware, high integration, good real-time and reliability of software. At the same time, it also needs to support a variety of business types. The design scheme proposed in this paper better grasps this trend. The device can not only maintain the operation process of traditional telephone for users, but also seamlessly connect with Wan, greatly reduce the communication cost and has strong practicability., Technology Zone
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