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Abstract : VoWLAN is a WLAN (Wireless Local Area Network)-based voice technology. It is a combination of WLAN and VoIP technologies. It belongs to a wireless VoIP technology. This technology makes it possible for people to make IP calls over the WLAN network anytime and anywhere.This terminal selects UBICOM's network processor IP2022 as the main controller of the terminal, and TI's TLV320 series AIC10 as the speech codec processor. The IP2022 is a RISC processor with an operating speed of 120 MIPS. In terms of voice processing, TI's DSP products are inexpensive and stable, and are widely used to meet the needs of voice at this stage. In addition, these devices have powerful computing and processing functions, and are easily expanded to meet future multimedia communication needs. The SIP protocol is the mainstream protocol for next-generation network (NGN) voice communications. It is open and can speed up product design cycles to meet the needs of society.
1 Introduction
1.1 VoWLAN Overview
VoWLAN is one of the emerging applications of WLAN. VoIP transmits voice signals over a data network; WLAN (Wireless Local Area Network), which wirelessly accesses the Internet through wireless access points. VoWLAN can be said to be an organic combination of the two, it can use the existing WLAN network to achieve wireless VoIP call capabilities, enterprise employees can use voice communication, access E-mail and other accesses outside the office space through VoWLAN Network resources, which increase the utilization of network resources and reduce the cost of calls, thereby saving the overall IT costs of the enterprise. For residential users, voice calls can also be reduced through VoIP calls connected to broadband 802.11 wireless networks. The data flow of the VoWLAN system is as follows: the voice signal is transmitted to the VoIP gateway through the access point AP (Access Point), and the gateway can be the original wired VoIP gateway, so that the voice data is transmitted between the IP network and the PBX (switch). In this way, the functions of traditional wired office and residential telephone can be realized in VoWLAN, and operators only need to add devices such as voice gateway and charging system to provide VoWLAN application for data communication and voice communication. Efficient combination.
1.2 SIP Overview
At present, there are three basic communication protocols involved in VoIP: H323 protocol, SIP protocol and MGCP protocol. SIP is called the Session Initiation Protocol and is one of the recommendations of the Internet Engineering Task Force (IETF). Compared with the H.323 protocol, the SIP protocol focuses on IP telephony as an application on the Internet. Compared with the actual application (such as FTP, E-mail, etc.), it increases the signaling and QoS requirements. The services they support are basically the same. Both use RTP as a protocol for media transport; SIP is a text-based protocol, similar to HTTP. Text-based coding means that the meaning of the header field is clear, such as From, To, and Subject. The superiority of this distributed, standard specification style, which requires little or no complicated documentation, has been well documented in past practice (now the popular mail protocol SMTP is such an example). The message body part of SIP is described by SDP. Each format in SDP is '=', which is also relatively simple. SIP conveniently supports supplementary services or intelligent services, as long as the SIP defined header field is fully utilized, and These services can be implemented by simply extending the SIP. For example, for call forwarding, you can implement this service by adding a Contact header field to the BYE request message and adding the third-party address you want to transfer to. For some intelligent services that are difficult to implement through extended header fields, service agents can be added to the architecture to provide supplementary services or interfaces with intelligent network devices. The SIP protocol draws on the design concepts of other Internet standards and protocols. It is relatively simple to follow the principles of simplicity, openness, compatibility and scalability that the Internet has consistently adhered to.
2 system hardware design
2.1 central controller
The central controller MCU uses UPUCOM's IP2022 network processor. It is a microprocessor designed for Internet-edge. Its main features: processing the protocol through software, rather than using hardware logic to implement the protocol, reducing the complexity of the chip; using the memory-to-memory operation for the arriving packets, without the need for large capacity, expensive on- Chip caches (cache), and packet buffers; simple design of real-time operating system, no memory-hungry requirement; Designer can design software to achieve a variety of Internet-edge applications; IP2022 Memory is divided into on-chip and Off-chip two parts, using Harvard architecture, separated data and code memories; 4k bytes on-chip data memory, 64k bytes of on-chip flash memory, 16k bytes of on-chip PRAM, external memory, extra Flash memory, used to put more code, such as the code of the embedded web server can be placed in the external flash, including some dll functions. And online firmware upgrades. SRAM is used to extend the PRAM space, such as buffer packets in SRAM.
2.2 Voice Module
This module uses the TLV320AIC10. The TLV320AIC10 is a new low-power ∑-Δ 16-bit A/D and D/A audio interface (AIC) chip that TI has introduced in recent years. It is controlled by five control registers. Among them, control register 1: software reset, DAC 16-bit or 15+1-bit mode selection and anti-aliasing filter, sampling filter, interpolation filter enable / bypass selection. Control Register 2: Determines the mode of operation and the sampling rate. Low power mode control, divider register control (determines the filter's clock frequency and sampling period). Control Register 3: Software Power Off, Analog and Digital Signal Feedback and Event Control Mode Selection; 16-bit or 15+1-bit mode selection for the ADC. Control Register 4: Input and Output Gain Control (implemented by controlling the input and output programmable gain amplifiers). The initialization of the AIC is mainly to set the four register parameters. The interface between the device and the microcontroller is easy to implement, and it is more convenient to develop and use. It is especially suitable for a variety of VoIP, cable modem, voice and telephony applications for voice transmission, identification and synthesis of low bit rate, high performance intensive devices.
2.3 network module
The wireless receiving and transmitting module selects the CF wireless network card slot. The CF card (CompactFlash) is a flash memory card introduced by SanDisk in 1994. The CF card has and is compatible with the PCMCIA-ATA function, which uses flash technology and is a stable storage solution that does not require a battery to maintain the data stored therein. For all saved data, CF cards are more secure and protective than traditional disk drives, and CF cards use only 5% of the capacity of small disk drives. These excellent conditions make most PDAs use a CF card as their preferred interface.
The wireless receiving and transmitting module of the system uses the WL-672F CF wireless network adapter. The wireless network adapter is a CompactFlash I type adapter that can be used with terminals equipped with a Type II slot. Use this adapter to keep E-mail and access server data when you are working on a mobile PDA. It uses a credit card design, provides an integrated antenna, can be interactively operated with all IEEE802.11b (DSSS) 2.4GHz compliant wireless network devices, and can be interactively operated via AP and wired Ethernet to support Ad-Hoc and The infrastructure communication method uses 128-bit WEP encryption to ensure network security.
Figure 1 Hardware frame diagram of VoWLAN voice terminal
After the connection is established, the user's analog voice is input through the AURXFP, AURXM, and AURXCP of the AIC10, and the analog signal is A/D converted to form a digital signal stream and then transmitted to the encoding module. The DSP in the encoding module compresses the voice data according to the system requirements, compresses it and puts it into the register of DOUT.
When receiving the voice data, the frame synchronization FS of the TLV320AIC10 is low level, and when the rising edge of the clock signal SCLK is switched, the system processor sends the voice data to the AIC10 through the DIN to decompress, forming a 64Kbps PCM stream, and feeding. The D/A performs digital-to-analog conversion, and finally the analog voice is output by the OUTP and OUTM of the AIC10.
3 VoWLAN software design
The system software is implemented based on the SIP protocol stack. Because the SIP protocol stack adopts a modular design idea, the system software can directly call the API provided by each module of the protocol stack. The message acquisition thread of the software application module is the result of real-time acquisition of the protocol stack processing (stored in the protocol stack message queue in the form of a message or event), and converts it into a message structure of the system application and stores it in the message queue of the application module. Figure 2 is a flow chart of the program of the software application module.
Figure 2 software application flow chart
The "application initialization" in FIG. 2 includes establishing a message reading thread, etc.; "SIP protocol stack initialization" includes establishing a protocol stack main thread, registering a callback function, initializing other modules of the protocol stack, and establishing a message queue; First, the logout operation is performed, and then the message read thread and the protocol stack main thread are terminated, and the resources occupied by the software application module and each module of the protocol stack are released.
The protocol stack module contains two modules, transaction management and dialog management. Similarly, the software application module also contains a similar management function, namely "call management". The function of call management is similar to the dialog management function in the protocol stack module, except that the call management focuses on interacting with the user, displaying information through the human-machine interface, prompting the user for the progress of the current call, and guiding the user to perform further operations.
The call has four states: "S_IDLE", "S_PROGRESS", "S_INCOMING", "S_CONNECTED", and the relationship between them is described by a finite state machine, as shown in Figure 3.
Figure 3 Call Management Finite State Machine
The "S_IDLE" state is what we usually call the standby state. When the user dials the SIP URI of the other party, the protocol stack obtains a temporary response message by sending an INVITE request message, and the call is in "S_PROGRESS". The arrow <1> in the figure indicates the state switch caused by the operation. In the "S_PROGRESS" state, if the other party refuses to accept the call, the status returns to "S_IDLE", as indicated by the arrow <2>; when the other party accepts the call, it is in the connected call state "S_CONNECTED", as indicated by the arrow <5>. In the "S_IDLE" state, the call request of the other party may also be received. When the UA application finds a new call request, it should switch the status to "S_INCOMING", as indicated by the arrow <3>; likewise, the user can refuse to accept the call and the status is switched back. "S_IDLE", as indicated by arrow <4>; when the user accepts the call request, the call status is "S_CONNECTED". In the "S_CONNECTED" state, the end call request of either party of the call will cause the status to return "S_IDLE" and end the call.
4 Conclusion
The cost of the terminal is low, and it is now able to register to the SIP test platform of the relevant manufacturer, and can complete the basic session function; at the same time, the LAN can be successfully registered in the static IP mode and perform session operations.
Innovation
The embedded implementation of the VoWLAN terminal of the SIP/SDP signaling protocol; the embedded implementation of the USER Agent in the VoWLAN terminal; the call of the VoWLAN terminal to the PC, the VoWLAN terminal to the VoWLAN terminal, and the VoWLAN terminal to the PSTN.
references
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