CN101026836A - Anti vocoder compressed end-to-end voice encryption device and method - Google Patents
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Abstract
抗声码器压缩的端到端语音加密装置与方法涉及全球移动通信系统中的端到端安全通信领域,具体是一种在GSM移动通信系统传统语音通道上实现的抗声码器压缩的端到端语音加密装置与方法。它由发送端加密模块(1)和接收端解密模块(2)组成。发送端加密模块(1)包括低码率压缩模块(4)、加密模块(5)、正交多载波调制模块(6)、第一载波生成器模块(7)、第一密钥管理协商模块(8)和第一同步模块(9);接收端解密模块(2)包括正交多载波解调模块(14)、第二载波生成器模块(15)、解密模块(16)、低码率解压缩模块(17)、第二密钥管理协商模块(18)和第二同步模块(19)。本发明通过GSM的语音通道进行端到端加密语音传输,具有时延低、安全性高、实现灵活方便等优点。
The end-to-end voice encryption device and method of anti-vocoder compression relate to the field of end-to-end secure communication in the global mobile communication system, specifically an end-to-end anti-vocoder compression implemented on the traditional voice channel of the GSM mobile communication system End-to-end voice encryption device and method. It consists of an encryption module (1) at the sending end and a decryption module (2) at the receiving end. The encryption module (1) at the sending end includes a low code rate compression module (4), an encryption module (5), an orthogonal multi-carrier modulation module (6), a first carrier generator module (7), and a first key management negotiation module (8) and the first synchronization module (9); the receiving end deciphering module (2) comprises an orthogonal multi-carrier demodulation module (14), a second carrier generator module (15), a deciphering module (16), a low code rate A decompression module (17), a second key management negotiation module (18) and a second synchronization module (19). The invention performs end-to-end encrypted voice transmission through the GSM voice channel, and has the advantages of low time delay, high security, flexible and convenient implementation, and the like.
Description
技术领域technical field
本发明涉及全球移动通信系统(GSM,Global System of Mobile communication)中的端到端安全通信领域,具体是一种在GSM移动通信系统传统语音通道上实现的抗声码器压缩的端到端语音加密装置与方法。The present invention relates to the field of end-to-end secure communication in the Global System of Mobile Communication (GSM, Global System of Mobile communication), in particular to an end-to-end voice that is compressed by an anti-vocoder implemented on the traditional voice channel of the GSM mobile communication system Encryption device and method.
背景技术Background technique
GSM是由国际电信联盟标准组织制定的基于时分多址方式的蜂窝移动通信标准,可提供话音、数据、短消息及无线上网等业务,在欧洲及我国率先得到应用,获得了极大成功。目前中国GSM网络已覆盖全国绝大部分城镇,成为国民经济各行各业重要的通信手段。GSM is a cellular mobile communication standard based on time-division multiple access mode formulated by the International Telecommunication Union Standardization Organization. It can provide services such as voice, data, short message and wireless Internet access. It was first applied in Europe and my country and achieved great success. At present, China's GSM network has covered most of the cities and towns in the country, and has become an important means of communication for all walks of life in the national economy.
随着GSM网络的飞速发展,大量涉及商业以及个人的敏感信息需要通过GSM手机传递。对于GSM手机通信这个开放的电子通信系统,GSM手机与基站间通过无线空中传输,通信的保密性差。GSM系统语音通信本身提供了一套安全管理功能,但其安全功能是不完整和不彻底的,存在着单向鉴权,A5加密算法脆弱和基站后语音透明传输等安全隐患;同时GSM本身系统的加密是由运营商控制的,系统实施复杂。为了保证通信的信息安全,需要采用额外加密的方法对传输的信息进行加密保护。无线通信网络一般采用空中接口加密和端到端加密两种方式。空中接口加密只对手机和基站间无线信道上的信息传输进行加密保护,数据容易被窃听和盗取。而端到端加密对从信源到信宿所有链路的全程进行加密保护,适合语音和数据的端到端的安全传输,具有较高的安全保密性。With the rapid development of the GSM network, a large amount of sensitive information related to business and personal needs to be transmitted through the GSM mobile phone. For the open electronic communication system of GSM mobile phone communication, GSM mobile phone and base station are transmitted through wireless air, and the confidentiality of communication is poor. The GSM system voice communication itself provides a set of security management functions, but its security functions are incomplete and incomplete, and there are security risks such as one-way authentication, weak A5 encryption algorithm and transparent transmission of voice behind the base station; at the same time, the GSM system itself The encryption is controlled by the operator, and the system implementation is complex. In order to ensure the security of communication information, it is necessary to use additional encryption methods to encrypt and protect the transmitted information. Wireless communication network generally adopts air interface encryption and end-to-end encryption in two ways. The air interface encryption only encrypts and protects the information transmission on the wireless channel between the mobile phone and the base station, and the data is easy to be eavesdropped and stolen. The end-to-end encryption encrypts and protects all links from the source to the destination, which is suitable for end-to-end secure transmission of voice and data, and has high security and confidentiality.
移动语音端到端加密需要在终端信源处理部分进行加密才能达到端到端的安全。移动通信可以承载语音和数据业务,所以语音传输有传统语音通道的语音传输和数据通道的语音传输(走电路交换链路数据通道或者GPRS上的IP协议通道),因而端到端的移动保密语音通信也有这两个通道的方法。语音通道加密的优点是延迟较小,并可与现有的移动网络的各种业务很好地融合(如电话卡和各种套餐),在现有的广大用户中有很好的普及性。数据通道加密的优点是由于避开了RPE-LTP声码器,因而易于实现保密话音通信。但数据通道加密语音传输具有以下的缺点:a)建立连接和运用自动重传机制造成的延时过大问题无法克服;b)数据通道在各区域移动网络之间存在互用性的问题;c)数据通道方式需要在运营商开通数据业务,而且两端不能使用现有的电话卡套餐等业务。The end-to-end encryption of mobile voice needs to be encrypted in the source processing part of the terminal to achieve end-to-end security. Mobile communication can carry voice and data services, so voice transmission includes voice transmission of traditional voice channel and voice transmission of data channel (via circuit switched link data channel or IP protocol channel on GPRS), so end-to-end mobile secure voice communication There are also methods for these two channels. The advantage of voice channel encryption is that the delay is small, and it can be well integrated with various services of the existing mobile network (such as phone cards and various packages), and it has good popularity among the existing users. The advantage of data channel encryption is that it is easy to realize confidential voice communication because it avoids the RPE-LTP vocoder. However, the data channel encrypted voice transmission has the following disadvantages: a) the problem of excessive delay caused by establishing a connection and using an automatic retransmission mechanism cannot be overcome; b) there is a problem of interoperability of the data channel between mobile networks in various regions; c ) The data channel method needs to open data services at the operator, and the existing phone card packages and other services cannot be used at both ends.
直接在语音通道上进行语音信号加密处理会使得加密后的信号不具有语音的特征,难以有效通过GSM系统的规则脉冲激励长时预测(RPE-LTP,Regular PulseExcited-Long Term Prediction)声码器。RPE-LTP声码器是一种基于混合参数压缩编码算法,通过提取输入语音的声音模型参数来进行压缩,然后传输压缩后的参数,而在接收端通过这些参数恢复出来原有的语音。一般接收端恢复出来的语音在语音特性上是跟原有语音基本一致的,但在波形上与原有语音有差异。正是这个原因限制了一般有线网络上的比特扰乱加密方法在GSM语音通道上使用。Performing voice signal encryption processing directly on the voice channel will make the encrypted signal not have the characteristics of voice, and it is difficult to effectively pass the Regular Pulse Excited-Long Term Prediction (RPE-LTP, Regular Pulse Excited-Long Term Prediction) vocoder of the GSM system. RPE-LTP vocoder is a compression coding algorithm based on mixed parameters. It compresses by extracting the sound model parameters of the input speech, and then transmits the compressed parameters, and restores the original speech through these parameters at the receiving end. Generally, the voice restored by the receiving end is basically the same as the original voice in terms of voice characteristics, but it is different from the original voice in waveform. It is for this reason that the bit scrambling encryption method on the general wired network is limited to be used on the GSM voice channel.
目前已有关于对手机的语音和数据进行加密的方法。如中国专利申请号为98108859.7的发明提出了一种新的通讯加密方法,但其主要是侧重于端到端密钥的协商过程。该方法的主要内容是:主叫方终端利用网络发来的随机数生成加密密钥,该加密密钥通过呼叫建立消息发送到被叫方终端,使两端用户在通讯时可以使用相同的密钥对通话信息进行加/解密。申请号为200410030771的发明描述了一种基于GSM数据通道上的IP语音包进行加密传输来实现手机语音加密的方法。申请号为200410021689.2的发明提供了一种GSM网络语音和数据业务的端到端加密方法。通过增加一个通用的外置终端保密装置,外置终端保密装置通过蓝牙技术与支持蓝牙数据通信功能的GSM移动台连接,可实现GSM网络话音和数据业务的端到端加密。申请号为200410024248的发明仅仅描述了一个在模拟通道上进行语音置乱加密的模型和方法,对具体的抗声码器压缩没有做深入的分析;申请号为200410013706的发明描述了仅适用于公共电话交换网络的采用信息隐藏的方法来进行加密语音传输的方法,没考虑在GSM上的声码器压缩的影响因素。At present, there are methods for encrypting voice and data of mobile phones. For example, the invention with the Chinese patent application No. 98108859.7 proposes a new communication encryption method, but it mainly focuses on the negotiation process of the end-to-end key. The main content of this method is: the calling party terminal uses the random number sent by the network to generate an encryption key, and the encryption key is sent to the called party terminal through the call setup message, so that the users at both ends can use the same encryption key when communicating. key to encrypt/decrypt call information. The invention with the application number of 200410030771 describes a method for encrypting and transmitting voice over IP based on the GSM data channel to realize voice encryption of mobile phones. The invention with application number 200410021689.2 provides an end-to-end encryption method for GSM network voice and data services. By adding a common external terminal security device, the external terminal security device is connected with the GSM mobile station supporting the Bluetooth data communication function through Bluetooth technology, and the end-to-end encryption of GSM network voice and data services can be realized. The invention whose application number is 200410024248 only describes a model and method for voice scrambling and encryption on an analog channel, without in-depth analysis of specific anti-vocoder compression; the invention whose application number is 200410013706 describes only applicable to public The method of encrypted voice transmission using information hiding in the telephone switching network does not consider the influence factors of the vocoder compression on the GSM.
从调研结果看,目前已有的专利大多都集中在对手机加密模型、方法、框架方面进行申请保护,而且多集中在数据通道传输语音的模式方面。针对语音通道抗声码器压缩的新型语音加密方法目前尚未查询到公开的专利。而本发明描述的一种GSM语音通道的抗声码器压缩的端到端加密传输装置与方法,通过将加密后的语音数据经正交多载波调制方式调制成可以通过GSM声码器的波形,实现的语音加密过程延迟低,语音可懂度高,语音质量好,能适应全GSM网络,不存在网络兼容问题,能适应运营商不同的语音资费收费政策。According to the survey results, most of the existing patents are focused on applying for protection of mobile phone encryption models, methods, and frameworks, and most of them focus on the mode of voice transmission through data channels. A new type of voice encryption method for voice channel anti-vocoder compression has not yet found any published patents. And the anti-vocoder compression end-to-end encryption transmission device and method of a kind of GSM voice channel described in the present invention, by the voice data after the encryption is modulated into the waveform that can pass through the GSM vocoder through the orthogonal multi-carrier modulation mode , the realized voice encryption process has low delay, high voice intelligibility, good voice quality, can adapt to the whole GSM network, there is no network compatibility problem, and can adapt to different voice tariff charging policies of operators.
发明内容Contents of the invention
技术问题:本发明的目的是通过GSM网络语音通道来实现抗声码器压缩的端到端语音加密装置与方法,提出了一种基于正交多载波调制的抗GSM RPE-LTP13Kbps全速率声码器压缩的端到端加密装置与方法,即抗声码器压缩的端到端语音加密装置与方法,结合低码率压缩模块、同步模块和密钥协商模块,在不需开通额外业务的前提下,实现简洁、快速、安全、低成本的GSM端到端安全语音通信装置。Technical problem: the object of the present invention is to realize the anti-vocoder compression end-to-end voice encryption device and method through the GSM network voice channel, and propose a kind of anti-GSM RPE-LTP13Kbps full-rate voice code based on orthogonal multi-carrier modulation The end-to-end encryption device and method for vocoder compression, that is, the end-to-end voice encryption device and method for anti-vocoder compression, combined with a low bit rate compression module, a synchronization module and a key agreement module, without the need to open additional services Under this condition, a simple, fast, safe and low-cost GSM end-to-end secure voice communication device is realized.
技术方案:抗声码器压缩的端到端语音加密装置的一个单向加密过程包括发送端加密模块和接收端解密模块;发送端加密模块包括低码率压缩模块、加密模块、正交多载波调制模块、第一载波生成器模块、第一密钥管理协商模块和第一同步模块;接收端解密模块包括正交多载波解调模块、第二载波生成器模块、解密模块、低码率解压缩模块、第二密钥管理协商模块和第二同步模块;其中,低码率压缩模块的输入端接原发送手机的A/D编码模块的输出端,低码率压缩模块和密钥管理协商模块的输出端接加密模块的输入端,加密模块、载波生成器模块、同步模块的输出端接正交多载波调制模块的输入端,正交多载波调制模块的输出端接原发送手机的第一RPE-LTP声码器的输入端;接收端解密模块由正交多载波解调模块、第二载波生成器、解密模块、低码率解压缩模块、第二密钥管理协商模块和第二同步模块组成,其中,正交多载波解调模块的输入端分别接第二载波生成器、第二同步模块和原接收手机的第二RPE-LTP声码器的输出端,正交多载波解调模块和第二密钥管理协商模块的输出端接解密模块的输入端,解密模块的输出端接低码率解压缩模块的输入端,低码率解压缩模块的输出端为接收手机的D/A解码模块的输入端。Technical solution: A one-way encryption process of an end-to-end voice encryption device that resists vocoder compression includes a sending end encryption module and a receiving end decryption module; the sending end encryption module includes a low bit rate compression module, an encryption module, and an orthogonal multi-carrier Modulation module, first carrier generator module, first key management negotiation module and first synchronization module; receiver decryption module includes orthogonal multi-carrier demodulation module, second carrier generator module, decryption module, low code rate solution Compression module, the second key management negotiation module and the second synchronization module; wherein, the input end of the low bit rate compression module is connected to the output end of the A/D encoding module of the original sending mobile phone, and the low bit rate compression module and the key management negotiation The output terminal of the module is connected to the input terminal of the encryption module, the output terminals of the encryption module, the carrier generator module and the synchronization module are connected to the input terminal of the quadrature multi-carrier modulation module, and the output terminal of the quadrature multi-carrier modulation module is connected to the first mobile phone of the original sending The input terminal of a RPE-LTP vocoder; the decryption module at the receiving end consists of an orthogonal multi-carrier demodulation module, a second carrier generator, a decryption module, a low code rate decompression module, a second key management negotiation module and a second The synchronous module is composed of, wherein, the input terminal of the orthogonal multi-carrier demodulation module is respectively connected with the output end of the second carrier generator, the second synchronous module and the second RPE-LTP vocoder of the original receiving mobile phone, and the orthogonal multi-carrier demodulator The output terminal of the tuning module and the second key management negotiation module is connected to the input terminal of the decryption module, the output terminal of the decryption module is connected to the input terminal of the low code rate decompression module, and the output terminal of the low code rate decompression module is the D of the receiving mobile phone. /A The input terminal of the decoding module.
采用正交多载波调制方法建立端到端的传输通道实现加密语音通信;从手机A/D量化编码模块过来的线性PCM数字语音是标准的8KHz采样,13bit量化的数据,这个数据经过低码率压缩模块压缩之后,首先变成为低码率的压缩码流;压缩后的码流通过加密模块进行加密处理后,送到正交多载波调制模块,通过与第一载波生成器模块生成的载波进行调制,形成能穿透声码器压缩的传输信号波形,然后送回到手机的基带处理芯片的第一RPE-LTP声码器进行RPE-LTP压缩编码,再送到信道编码模块进行信道编码等其他操作,最后通过手机天线送出;在手机接收端,从天线下来的信号经过信道解码模块、第二RPE-LTP声码器处理后,首先送到接收端解密模块的正交多载波解调模块上,与本地的第二载波生成器模块一起进行解调处理;解调出来的信号经过解密模块解密,送到低码率解压缩模块,恢复出来线性PCM的8KHz采样,13bit量化数字语音的数据,最后送到手机里面的D/A解码模块,形成解密语音流。Use the orthogonal multi-carrier modulation method to establish an end-to-end transmission channel to realize encrypted voice communication; the linear PCM digital voice from the A/D quantization coding module of the mobile phone is standard 8KHz sampling, 13bit quantized data, and this data is compressed at a low bit rate After the module is compressed, it first becomes a compressed code stream with a low code rate; the compressed code stream is encrypted by the encryption module, and then sent to the orthogonal multi-carrier modulation module, which is processed by the carrier wave generated by the first carrier generator module. Modulate to form a transmission signal waveform that can pass through the vocoder compression, and then send it back to the first RPE-LTP vocoder of the baseband processing chip of the mobile phone for RPE-LTP compression coding, and then send it to the channel coding module for channel coding and other At the receiving end of the mobile phone, the signal from the antenna is processed by the channel decoding module and the second RPE-LTP vocoder, and then sent to the orthogonal multi-carrier demodulation module of the receiving end decryption module , together with the local second carrier generator module for demodulation processing; the demodulated signal is decrypted by the decryption module and sent to the low bit rate decompression module to restore the linear PCM 8KHz sampling and 13bit quantized digital voice data. Finally, it is sent to the D/A decoding module in the mobile phone to form a decrypted voice stream.
低码率压缩模块和低码率解压缩模块配对使用,将输入的8KHz,13bit的PCM语音用AMBE编码算法压缩成低于2.4K的码流,或者将压缩后的码流用AMBE解码算法解压缩为8KHz,13bit的PCM语音信号。加密模块和解密模块配对使用,模块内的加解密算法采用增强加密标准的AES数字加密算法,或者是基于比特流加密的RC4算法。The low bit rate compression module and the low bit rate decompression module are used in pairs to compress the input 8KHz, 13bit PCM voice into a bit stream lower than 2.4K with the AMBE encoding algorithm, or decompress the compressed bit stream with the AMBE decoding algorithm 8KHz, 13bit PCM voice signal. The encryption module and the decryption module are used in pairs. The encryption and decryption algorithm in the module adopts the AES digital encryption algorithm of the enhanced encryption standard, or the RC4 algorithm based on bit stream encryption.
第一载波生成器模块、第二载波生成器模块和正交多载波调制模块、正交多载波解调模块结合在一起来实现对压缩后语音码流的正交多载波调制传输和解调过程;这个正交多载波传输过程有两种实现模式;一种是与第一载波生成器模块、或第二载波生成器模块生成的载波直接调制解调来完成;另外一种方法就是采用正交频分复用算法来快速实现。The first carrier generator module, the second carrier generator module, the orthogonal multi-carrier modulation module, and the orthogonal multi-carrier demodulation module are combined to realize the orthogonal multi-carrier modulation transmission and demodulation process of the compressed voice code stream ; This orthogonal multi-carrier transmission process has two implementation modes; one is to complete the direct modulation and demodulation with the carrier generated by the first carrier generator module or the second carrier generator module; the other method is to use orthogonal Frequency division multiplexing algorithm to achieve fast.
采用正交多载波调制方法建立端到端的传输通道时,采用正交频分复用来实现快速正交多载波调制和解调,正交多载波采用的子载波频率低于1500Hz,各子载波上调制方式采用QPSK调制方式。When the orthogonal multi-carrier modulation method is used to establish an end-to-end transmission channel, orthogonal frequency division multiplexing is used to realize fast orthogonal multi-carrier modulation and demodulation. The up modulation mode adopts QPSK modulation mode.
采用正交频分复用来实现快速正交多载波调制和解调时,首先将输入的码流做串并转换,然后填充到指定的载波点上,而其他载波点上填0值,同时为保证快速傅立叶变换FFT运算出来的是实数,需要保证做快速傅立叶变换FFT和快速反傅立叶变换IFFT之前的数据基于中心点前后共轭偶对称;采用快速反傅立叶变换IFFT和快速傅立叶变换FFT的变换来实现正交多载波的调制和解调。When adopting OFDM to realize fast orthogonal multi-carrier modulation and demodulation, the input code stream is first converted to serial and parallel, and then filled to the specified carrier point, while other carrier points are filled with 0 values, and at the same time In order to ensure that the fast Fourier transform FFT calculation is a real number, it is necessary to ensure that the data before the fast Fourier transform FFT and the fast inverse Fourier transform IFFT are based on the front and rear conjugate even symmetry of the center point; the transformation of the fast inverse Fourier transform IFFT and the fast Fourier transform FFT is used To realize the modulation and demodulation of orthogonal multi-carrier.
加密语音通信两端需要采用第一密钥管理协商模块、第二密钥管理协商模块来实现通信两端的密钥同步和密钥更新管理;密钥交换过程由通信两方经迪夫-哈夫曼密钥协商机制来完成。Both ends of the encrypted voice communication need to use the first key management negotiation module and the second key management negotiation module to realize key synchronization and key update management at both ends of the communication; Key agreement mechanism to complete.
加密语音通信两端还需要同步模块来实现加密帧的同步;在每一帧开头加上一个训练同步序列,该同步序列在时域中包含前后两个相同的部分;取一个同步符号包含两个周期的正弦波或者方波,正弦波或者方波的频率为300Hz到500Hz之间;在接收端,对接收到的前后两半的样值点做相关运算并累加,得到的峰值点就是同步起始点;发送端对每个加密后的正交多载波帧加上同步序列,而在接收端,相应的同步序列被去掉。The two ends of the encrypted voice communication also need a synchronization module to realize the synchronization of the encrypted frame; a training synchronization sequence is added at the beginning of each frame, and the synchronization sequence contains two identical parts before and after in the time domain; a synchronization symbol contains two Periodic sine wave or square wave, the frequency of sine wave or square wave is between 300Hz and 500Hz; at the receiving end, the correlation calculation is performed on the received sample points of the two halves and accumulated, and the peak point obtained is the synchronous starting point. The starting point; the sending end adds a synchronization sequence to each encrypted orthogonal multi-carrier frame, and at the receiving end, the corresponding synchronization sequence is removed.
从手机A/D量化编码模块过来的线性PCM数字语音是标准的8KHz采样,13bit量化的数据,这个数据经过低码率压缩模块压缩之后,首先变成为低码率的压缩码流,一般低于2.4Kbps。这样在20ms的GSM语音内,用小于48bit的码就可以有效表示语音信息了。压缩后的码流通过加密模块进行加密处理后,送到正交多载波调制模块,通过与载波生成器生成的载波进行调制,形成能穿透声码器压缩的传输信号波形,然后送回到手机的基带处理芯片的RPE-LTP声码器进行RPE-LTP压缩编码,再送到信道编码模块进行信道编码等其他操作,最后通过手机天线送出。The linear PCM digital voice from the mobile A/D quantization coding module is standard 8KHz sampling and 13bit quantized data. After the data is compressed by the low bit rate compression module, it first becomes a low bit rate compressed bit stream. at 2.4Kbps. In this way, within the 20ms GSM voice, the voice information can be effectively represented with a code smaller than 48bit. After the compressed code stream is encrypted by the encryption module, it is sent to the orthogonal multi-carrier modulation module, and modulated with the carrier wave generated by the carrier generator to form a transmission signal waveform that can pass through the vocoder compression, and then sent back to the The RPE-LTP vocoder of the mobile phone's baseband processing chip performs RPE-LTP compression coding, and then sends it to the channel coding module for channel coding and other operations, and finally sends it out through the mobile phone antenna.
在手机接收端,从天线下来的信号经过信道解码模块、RPE-LTP声码器等处理后,首先送到接收端解密模块的正交多载波解调模块上,与本地的载波生成器一起进行解调处理。解调出来的信号经过解密模块解密,送到低码率解压缩模块,恢复出来线性PCM的8KHz采样,13bit量化数字语音的数据,最后送到手机里面的D/A解码模块,形成解密语音流。在手机发送接收两端,对应的密钥管理协商模块和同步模块完成通话两端的密钥协商和加密帧同步的过程。At the receiving end of the mobile phone, after the signal from the antenna is processed by the channel decoding module and the RPE-LTP vocoder, it is first sent to the orthogonal multi-carrier demodulation module of the decryption module at the receiving end, and is carried out together with the local carrier generator. demodulation processing. The demodulated signal is decrypted by the decryption module and sent to the low bit rate decompression module to recover the linear PCM 8KHz sampling and 13bit quantized digital voice data, and finally sent to the D/A decoding module in the mobile phone to form a decrypted voice stream . At the sending and receiving ends of the mobile phone, the corresponding key management negotiation module and synchronization module complete the process of key negotiation and encrypted frame synchronization at both ends of the call.
低码率压缩模块和低码率解压缩模块配对使用,采用压缩码率较低的AMBE编码算法,将输入的8KHz,13bit的PCM语音压缩成低于2.4Kbps的码流,或者将压缩后的码流解压为8KHz,13bit的PCM语音信号。AMBE编码算法是多带激励编码算法的一种,能实现高效的声音压缩编码。加密模块和解密模块必须配对使用。通常采用增强加密标准(AES)数字加密算法,或采用RC4比特流加密算法。AES算法是分组加密算法的一种,加密强度高。RC4算法是流加密算法的一种,加密速度快。The low bit rate compression module and the low bit rate decompression module are used in pairs, using the AMBE encoding algorithm with a lower compression bit rate to compress the input 8KHz, 13bit PCM voice into a bit stream lower than 2.4Kbps, or compress the compressed The code stream is decompressed into 8KHz, 13bit PCM voice signal. The AMBE coding algorithm is a kind of multi-band excitation coding algorithm, which can realize high-efficiency sound compression coding. Encryption module and decryption module must be used in pairs. Usually, the Enhanced Encryption Standard (AES) digital encryption algorithm is used, or the RC4 bit stream encryption algorithm is used. The AES algorithm is a kind of block encryption algorithm with high encryption strength. The RC4 algorithm is a kind of stream encryption algorithm, and the encryption speed is fast.
载波生成器模块与正交多载波调制模块和解调模块结合在一起实现对压缩后语音码流的正交多载波调制传输和解调过程。这个正交多载波传输过程有两种实现模式。一种是用载波生成器模块生成的载波直接调制解调来完成。载波生成器模块生成对应的各载波信号。根据GSM通话的语音频率范围(50Hz到3400Hz),载波生成器输出相关频点的频率,并在此频率上调制解调,需要保证各载波之间正交。通过我们的实验验证,低于1500Hz的正弦波通过四相位相移键控的QPSK调制携带压缩后的比特流数据透过声码器传输误码率可以小于10-3。当采用较高频率进行调制解调的时候,传输的误码率会变高。正交多载波的子载波数量与所用频率局限,正交频率的间隔,以及所需调制数据的传输速率都是有关的。子载波的个数乘以正交频率间隔即为当前调制解调使用的最高频率,应保持在误码率较低的频率范围内(如大致1500Hz以内),但同时子载波的数量会在一定程度上影响传输的码率,所以子载波的数量需要一个合适的数值。设子载波个数为n,数据的速率为Rb b/s,信号的最高频率为Fm。假设子载波采用PSK的相位调制方式,每个子载波可以调制m bit的数据。设起始频率等于频率的间隔,则调制载波信号的最高频率可推导出来为:The carrier generator module is combined with the orthogonal multi-carrier modulation module and the demodulation module to realize the orthogonal multi-carrier modulation transmission and demodulation process for the compressed voice code stream. There are two implementation modes for this orthogonal multi-carrier transmission process. One is to use the direct modulation and demodulation of the carrier generated by the carrier generator module. The carrier generator module generates corresponding carrier signals. According to the voice frequency range of the GSM call (50Hz to 3400Hz), the carrier generator outputs the frequency of the relevant frequency point, and modulates and demodulates on this frequency, and it is necessary to ensure that the orthogonality between the carriers is ensured. Through our experiments, the sine wave below 1500 Hz is modulated by quadrature phase shift keying QPSK to carry the compressed bit stream data through the vocoder and the bit error rate can be less than 10 -3 . When a higher frequency is used for modulation and demodulation, the bit error rate of the transmission will become higher. The number of sub-carriers of the orthogonal multi-carrier is related to the frequency limitation used, the interval of the orthogonal frequency, and the transmission rate of the required modulated data. The number of subcarriers multiplied by the orthogonal frequency interval is the highest frequency used by the current modulation and demodulation, which should be kept in the frequency range with a low bit error rate (such as within approximately 1500Hz), but at the same time the number of subcarriers will be within a certain range. To a certain extent, it affects the bit rate of transmission, so the number of subcarriers needs an appropriate value. Assume that the number of subcarriers is n, the data rate is R b b/s, and the highest signal frequency is F m . Assuming that the subcarrier adopts the phase modulation mode of PSK, each subcarrier can modulate m bit data. Assuming that the starting frequency is equal to the frequency interval, the highest frequency of the modulated carrier signal can be derived as:
Fm=Rb/mHzF m =R b /mHz
由上式可以看出,最高频率与输入数据速率需求和每个子载波上调制的比特数有直接关系,而与子载波的数量并没有直接的关系。也就是说我们想要得到较低误码率的最高频率,当输入数据速率需求确定之后,增加子载波上携带的比特数可以很明显的降低正交多载波子载波的最高频率。每个子载波可调制的比特数与声码器的系统特性有关,通过实验证明,经过声码器的正弦波形最多对2比特的QPSK调制有较小的误比特率。所以,在进入声码器之前,我们将语音压缩至小于2.4kb/s,采用24个正交子载波进行QPSK调制,可以控制使用的最高频率小于1500Hz,实现较低的误码率。It can be seen from the above formula that the highest frequency is directly related to the input data rate requirement and the number of bits modulated on each subcarrier, but not directly related to the number of subcarriers. That is to say, we want to obtain the highest frequency with a lower bit error rate. When the input data rate requirement is determined, increasing the number of bits carried on the subcarrier can obviously reduce the highest frequency of the orthogonal multi-carrier subcarrier. The number of bits that can be modulated by each subcarrier is related to the system characteristics of the vocoder. It is proved by experiments that the sinusoidal waveform passing through the vocoder has a smaller bit error rate for QPSK modulation of 2 bits at most. Therefore, before entering the vocoder, we compress the voice to less than 2.4kb/s, and use 24 orthogonal subcarriers for QPSK modulation, which can control the highest frequency used to be less than 1500Hz to achieve a lower bit error rate.
正交多载波调制解调过程实现的另外一种方法就是采用正交频分复用(OFDMOrthogonal Frequency Division Multiplexing)算法来快速实现。采用快速反傅立叶变换IFFT和快速傅立叶变换FFT的变换来实现正交多载波的调制和解调。Another way to realize the orthogonal multi-carrier modulation and demodulation process is to use the Orthogonal Frequency Division Multiplexing (OFDM Orthogonal Frequency Division Multiplexing) algorithm to quickly realize. Modulation and demodulation of orthogonal multi-carriers are realized by transforming fast inverse Fourier transform IFFT and fast Fourier transform FFT.
正交频分复用OFDM调制方法是多载波调制技术的一种。它的基本思想是把数据流串并变换为N路速率较低的子数据流,用它们分别去调制N路子载波后并行传输。因子数据流的速率是原来的1/N,即符号周期扩大为原来的N倍,这样OFDM就把一个宽带频率选择性信道划分成了N个窄带平坦衰落信道,从而具有很强的抗无线信道多径衰落和抗脉冲干扰的能力,特别适合于高速无线数据传输。OFDM选择时域相互正交的子载波,他们虽然在频域相互混叠,却仍能在接收端被分离出来。Orthogonal frequency division multiplexing OFDM modulation method is a kind of multi-carrier modulation technology. Its basic idea is to serially convert data streams into N sub-data streams with lower rates, use them to modulate N sub-carriers respectively, and then transmit them in parallel. The rate of the factor data flow is the original 1/N, that is, the symbol period is expanded to N times of the original, so that OFDM divides a wideband frequency selective channel into N narrowband flat fading channels, which has a strong resistance to wireless channels The ability of multipath fading and anti-pulse interference is especially suitable for high-speed wireless data transmission. OFDM selects subcarriers that are orthogonal to each other in the time domain. Although they alias each other in the frequency domain, they can still be separated at the receiving end.
在正交多载波调制模块中,可采用OFDM调制来完成。OFDM发送端调制原理如下:速率为Rb bps的串行比特流,根据反傅里叶变换IFFT的大小N(如128的IFFT),则首先进行串并转换,变换成N排的并行流,每排并行流代表一个正交的子载波。然后,根据前面低码率压缩模块压缩后的码流(如2.4Kbps,则每20ms有48bit信息),首先填充对应子载波的那一排的码。承载这个有效信息的载波仍需要控制在GSM语音频谱50Hz到3400Hz之间,而且要保证QPSK调制能通过声码器则子载波的最高频率需要小于1500Hz。除承载有效压缩语音信息的子载波外,其他排子载波上的码一般填充0值,需要注意的是,为保证IFFT出来的是实数,128点的IFFT按照中心点做镜像处理,也就是说后半数据是前半数据是共轭偶对称。这样,串并转换好之后的码进行QPSK调制,然后经过反傅里叶变换IFFT。最后再将变换之后的信号叠加,并串转换。形成调制后的波形,加上同步序列之后,就可以送到声码器进行传输。In the orthogonal multi-carrier modulation module, OFDM modulation can be used to complete. The modulation principle of the OFDM sending end is as follows: the serial bit stream with a rate of R b bps, according to the size N of the inverse Fourier transform IFFT (such as 128 IFFT), first performs serial-to-parallel conversion and converts it into N rows of parallel streams, Each row of parallel streams represents an orthogonal subcarrier. Then, according to the code stream compressed by the previous low code rate compression module (for example, 2.4Kbps, there is 48bit information every 20ms), the code of the row corresponding to the subcarrier is firstly filled. The carrier carrying this effective information still needs to be controlled within the GSM voice spectrum between 50Hz and 3400Hz, and to ensure that QPSK modulation can pass through the vocoder, the highest frequency of the subcarrier needs to be less than 1500Hz. Except for subcarriers carrying effectively compressed voice information, the codes on other subcarriers are generally filled with 0 values. It should be noted that in order to ensure that the IFFT results are real numbers, the 128-point IFFT is mirrored according to the center point, that is to say The second half of the data is the first half of the data is conjugate even symmetry. In this way, the serial-to-parallel converted code is subjected to QPSK modulation, and then undergoes inverse Fourier transform IFFT. Finally, the transformed signals are superimposed and converted to parallel and serial. After forming the modulated waveform and adding the synchronization sequence, it can be sent to the vocoder for transmission.
在OFDM解调端,从声码器出来的OFDM调制信号,首先经过傅里叶变换FFT处理进行。然后根据调制时候的码流对应表,从变换处理后的信息中提取出来有效的被压缩后的语音信息,经并串转换后,送到低码率解码模块进行解码处理,就可以解码出语音来。At the OFDM demodulation end, the OFDM modulated signal from the vocoder is firstly processed by Fourier transform FFT. Then, according to the code stream correspondence table at the time of modulation, the effective compressed voice information is extracted from the converted information, and after parallel-to-serial conversion, it is sent to the low bit rate decoding module for decoding processing, and the voice can be decoded Come.
加密通话两端需要采用密钥管理协商模块来实现通信两端的密钥同步和密钥更新管理。密钥交换的过程可以由通信两方通过迪夫-哈夫曼(Diffie-Hellman)密钥协商机制来完成。Both ends of an encrypted call need to use a key management negotiation module to realize key synchronization and key update management at both ends of the communication. The key exchange process can be completed by the two communicating parties through the Diffie-Hellman (Diffie-Hellman) key agreement mechanism.
加密语音通信两端还需要同步模块来实现加密帧的同步。同步是任何一个数字通信系统中的重要环节,没有准确的同步算法,是无法准确可靠的收发数据的。正交多载波调制解调系统对同步的要求非常高,符号定时的不准确,将会直接影响到整个系统的性能。符号定时的目的是为了找到解调窗的起始位置,采用特殊的训练同步序列来进行符号定时。Both ends of encrypted voice communication also need a synchronization module to realize the synchronization of encrypted frames. Synchronization is an important link in any digital communication system. Without an accurate synchronization algorithm, it is impossible to send and receive data accurately and reliably. Orthogonal multi-carrier modulation and demodulation systems have very high requirements for synchronization, and inaccurate symbol timing will directly affect the performance of the entire system. The purpose of symbol timing is to find the starting position of the demodulation window, and a special training synchronization sequence is used for symbol timing.
本发明中,每一帧开头加上一个训练同步序列。该同步序列在时域中包含前后两个相同的部分。取一个正弦波信号或者方波信号作为同步符号的一半,相当于一个同步符号包含两个周期的正弦波或者方波,正弦波或者方波的频率一般在300Hz到500Hz之间。在接收端,对接收到的前后两半样值点做相关运算并累加,峰值点就是同步开始点。发送端对每个加密后的正交多载波帧都加上同步符号,而在接收端,必须将同步符号去掉之后再进行数据的解调。In the present invention, a training synchronization sequence is added at the beginning of each frame. The synchronization sequence contains two identical parts before and after in the time domain. Taking a sine wave signal or a square wave signal as half of the synchronization symbol is equivalent to a sine wave or square wave containing two cycles in a synchronization symbol, and the frequency of the sine wave or square wave is generally between 300Hz and 500Hz. At the receiving end, correlation calculations are performed on the received two halves of sample points and accumulated, and the peak point is the synchronization start point. The sending end adds synchronization symbols to each encrypted orthogonal multi-carrier frame, but at the receiving end, the synchronization symbols must be removed before data demodulation.
本发明描述的抗声码器压缩的端到端加密方法,所有的模块可以在DSP上通过软件实现,最后形成单芯片的加解密模块。考虑到低码率压缩模块和低码率解压缩模块的复杂度,也可以将低码率压缩模块和低码率解压缩模块采用现成的芯片来替代,配合DSP,用两个芯片完成GSM上抗声码器的端到端语音加密传输过程。In the end-to-end encryption method for anti-vocoder compression described in the present invention, all modules can be implemented on the DSP through software, and finally form a single-chip encryption and decryption module. Considering the complexity of the low-bit-rate compression module and the low-bit-rate decompression module, the low-bit-rate compression module and the low-bit-rate decompression module can also be replaced by ready-made chips. End-to-end speech encryption transmission process against vocoder.
本发明中低码率压缩模块采用2.4Kbps码率的压缩算法,另外也可以采用更低的如1.2Kbps到1.6Kbps码率的压缩算法,加上前向纠错算法后,传输加密语音的误码率更低,语音的效果会更好。The low code rate compression module of the present invention adopts the compression algorithm of 2.4Kbps code rate, also can adopt lower compression algorithm such as 1.2Kbps to 1.6Kbps code rate in addition, after adding the forward error correction algorithm, the error of transmitting encrypted voice The lower the bit rate, the better the voice effect.
有益效果:本发明的创新点在于采用正交多载波调制传输方式,结合低码率语音压缩、数据加密、同步和密钥管理协商模块,实现基于语音通道的端到端加密。其的优势在于:Beneficial effects: the innovation of the present invention lies in the use of orthogonal multi-carrier modulation transmission mode, combined with low bit rate voice compression, data encryption, synchronization and key management negotiation modules, to realize end-to-end encryption based on voice channel. Its advantages are:
1)在GSM的语音信道上传输加密语音,方便快捷,不需要额外开通新业务。1) It is convenient and quick to transmit encrypted voice on the voice channel of GSM, and does not need to open additional new services.
2)真正的端到端加密过程,可以通过手机终端到手机终端或者有线终端。2) The real end-to-end encryption process can pass from mobile terminal to mobile terminal or wired terminal.
3)处理延迟小,小于100ms。声音可懂度高,保密强度强。3) The processing delay is small, less than 100ms. High sound intelligibility and strong confidentiality.
4)抗压缩,抗滤波,抗重采样。4) Anti-compression, anti-filtering, anti-resampling.
5)适应性强,通过适当的修改可以应用到CDMA和3G系统中。5) Strong adaptability, can be applied to CDMA and 3G systems through appropriate modification.
6)资费可以利用原有的语音资费优惠,简单便宜。6) The tariff can use the original voice tariff discount, which is simple and cheap.
7)不影响核心网络体系结构,对运营商影响小。7) It does not affect the core network architecture and has little impact on operators.
附图说明Description of drawings
下面结合附图和实施例对本发明进一步说明。The present invention will be further described below in conjunction with the accompanying drawings and embodiments.
图1是抗声码器压缩的端到端语音加密装置与方法总框图,描述了一个单向端到端加解密过程的各模块连接图。FIG. 1 is a general block diagram of an end-to-end speech encryption device and method against vocoder compression, describing a connection diagram of each module in a one-way end-to-end encryption and decryption process.
图2是插入同步之后的正交多载波调制时域波形图。Fig. 2 is a time-domain waveform diagram of orthogonal multi-carrier modulation after synchronization is inserted.
图3是与载波生成器模块结合的正交多载波调制示意图。Fig. 3 is a schematic diagram of quadrature multi-carrier modulation combined with a carrier generator module.
图4是基于OFDM的正交多载波调制示意图。Fig. 4 is a schematic diagram of OFDM-based orthogonal multi-carrier modulation.
具体实施方式Detailed ways
为进一步说明本发明的上述目的、技术方案和效果,以下通过实施例结合上述各图对本发明进行详细的描述。In order to further illustrate the above-mentioned purpose, technical solution and effect of the present invention, the present invention will be described in detail below through examples in conjunction with the above-mentioned figures.
抗声码器压缩的端到端语音加密装置的一个单向加密过程由发送端加密模块和接收端解密模块组成。它们分别与发送端手机和接收端手机相结合,实现单向通信语音的加解密处理过程。双向的语音加密则需要在安全通话的发送端手机上增加解密模块和接收端手机增加加密模块。发送端加密模块由低码率压缩模块、加密模块、正交多载波调制模块、载波生成器模块、密钥管理协商模块和同步模块组成。接收端解密模块由正交多载波解调模块、载波生成器、解密模块、低码率解压缩模块、密钥管理协商模块和同步模块组成。A one-way encryption process of the anti-vocoder compression end-to-end voice encryption device is composed of a sending end encryption module and a receiving end decryption module. They are respectively combined with the mobile phone at the sending end and the mobile phone at the receiving end to realize the encryption and decryption process of the one-way communication voice. Two-way voice encryption needs to add a decryption module on the mobile phone at the sending end of the secure call and an encryption module at the mobile phone at the receiving end. The encryption module at the sending end is composed of a low code rate compression module, an encryption module, an orthogonal multi-carrier modulation module, a carrier generator module, a key management negotiation module and a synchronization module. The receiver decryption module is composed of an orthogonal multi-carrier demodulation module, a carrier generator, a decryption module, a low code rate decompression module, a key management negotiation module and a synchronization module.
抗声码器压缩的端到端语音加密装置与方法总框图如图1所示。分为发送端加密模块和接收端解密模块。加密过程的处理步骤如下:The overall block diagram of the end-to-end speech encryption device and method against vocoder compression is shown in FIG. 1 . It is divided into a sending end encryption module and a receiving end decryption module. The processing steps of the encryption process are as follows:
a)从手机A/D量化编码模块3取下来的8KHz,13bit PCM格式语音信号送入发送端加密模块1的低码率压缩模块4。这个PCM码流是同步时钟驱动的信号,由时钟线、字同步线和接收数据线组成。反向传输的话还有发送数据线。低码率压缩模块4将这个104Kbps的信号经AMBE压缩算法压缩为小于2.4Kbps的码流,可以采用单独低码率压缩芯片实现或者通过DSP实现。语音处理一般按照20ms为一帧处理,小于2.4Kbps是为了保证20ms时间内压缩后的有效信息小于48bit,以保证后面的正交多载波调制模块只采用较低误码率的低频段载波来完成调制。当压缩码率小于2.4Kbps时,如1.2Kbps或1.6Kbps时,可以加上前向纠错算法来改善后面正交调制传输的误码率性能。a) The 8KHz that is taken off from the mobile phone A/D
b)低码率编码模块4压缩后的码流,送入加密模块5进行加密操作。采用增强加密标准AES算法进行加密,或者采用RC4比特流加密算法来进行加密。b) The code stream compressed by the low code
c)加密处理后的码流送入正交多载波调制模块6,与载波生成器模块7一起进行正交多载波的调制。另外也可以对送入的码流作基于OFDM的正交多载波调制处理。选择的子载波频率要小于1500Hz,相互之间正交。调制方式采用QPSK调制进行。c) The encoded code stream is sent to the orthogonal
d)正交多载波调制好的波形最后送回到手机内的RPE-LTP声码器10,然后送到信道编码模块11等其他部分进行信道处理,最后从空中发送出去。d) The waveform modulated by the orthogonal multi-carrier is finally sent back to the RPE-
e)密钥管理协商模块8和同步模块完成加密通信两端的密钥协商、密钥分发管理和加密帧同步的功能。密钥交换的过程由通信两方通过迪夫-哈夫曼密钥协商机制来完成。同步则通过在调制后的波形前面加上相应的同步序列来进行每加密帧的同步。e) The key management and
解密端的处理步骤如下:The processing steps at the decryption end are as follows:
a)从手机空中接收下来的信号经过信道解码模块12等处理后,再经过RPE-LTP声码器13解码后,变成了加密调制过的8KHz,13bit PCM码流。将这个码流送入到正交多载波解调模块14中。a) After the signal received from the mobile phone in the air is processed by the
b)正交多载波解调模块14与载波生成器模块15相结合,进行对应于正交多载波调制处理的对应解调处理。或者通过进行对应OFDM解调处理。最后解调出来被加密的码流。b) The orthogonal
c)解调出来的被加密码流经过解密模块16解密,得到解密的被压缩码流信息。c) The demodulated encrypted code stream is decrypted by the
d)低码率解码模块17将送过来的解密后的码流进行解码操作,恢复出来原来的8KHz,13bit的语音PCM信号,这个信号再送回到手机内的D/A解码模块,最后通过喇叭恢复出发送端的声音。d) The low bit
在发送加密端,经过正交多载波调制后的信号,与同步信号一起组成一个加密帧结构,传送出去。如图2所示,在一个20ms为处理单位的语音帧里面,划出前面4ms的时间填充同步信号,可采用填充两个500Hz的正弦波或者方波,幅度略大于后面信号的统计最大幅值,而20ms后面的16ms填充的是正交多载波调制后的数据。在解调的时候,通过将后2ms的波形与前2ms的波形进行滑动相关运算,最后求得一个20ms范围内的一个最高峰值就是一个同步头开始。而从这个同步头开始4ms之后的就是传输的调制数据。At the sending encryption end, the signal modulated by the orthogonal multi-carrier forms an encrypted frame structure together with the synchronous signal and is transmitted. As shown in Figure 2, in a voice frame with a processing unit of 20ms, the first 4ms time is drawn to fill the synchronization signal, and two 500Hz sine waves or square waves can be used to fill in, and the amplitude is slightly larger than the statistical maximum amplitude of the latter signal. , and the 16ms after 20ms are filled with data after orthogonal multi-carrier modulation. During demodulation, the waveform of the last 2ms and the waveform of the first 2ms are subjected to sliding correlation calculation, and finally a highest peak value in the range of 20ms is obtained, which is the start of a synchronization head. And after 4ms from the synchronization header is the transmitted modulated data.
本发明核心是正交多载波调制和解调的处理过程。在与载波生成器生成的正交子载波进行调制处理时,需要按照图3所示的步骤进行:The core of the invention is the process of orthogonal multi-carrier modulation and demodulation. When performing modulation processing with the orthogonal subcarriers generated by the carrier generator, it is necessary to follow the steps shown in Figure 3:
a)首先按照前面压缩编码的码率速率要求,按照QPSK调制的方式,选择对应的载波数。由于在一个20ms的时间内,16ms是设计用来传输调制信号的。则正交子载波的载波间隔为62.5Hz。考虑选择的载波频率需要小于1500Hz(调制正弦波误码率较低)。所以首先确定起始子载波频率,然后确定其他的子载波,如设输入的码流需求为2.4Kbps。表示在20ms里面有48bit信息。当采用QPSK时。需要用24个子载波,则选择62.5Hz到1500Hz之间间隔为62.5Hz的24个子载波作为有效载波。当前面码率要求小于2.4Kbps时,则需要采用更少的载波进行调制传输。a) First, select the corresponding number of carriers according to the code rate requirement of the previous compression coding and the QPSK modulation mode. Since in a 20ms period, 16ms is designed to transmit the modulated signal. Then the carrier spacing of the orthogonal subcarriers is 62.5 Hz. Consider that the selected carrier frequency needs to be less than 1500Hz (modulated sine wave bit error rate is low). So first determine the initial subcarrier frequency, and then determine other subcarriers, for example, set the input code stream requirement to 2.4Kbps. It means that there is 48bit information in 20ms. When using QPSK. If 24 sub-carriers are required, 24 sub-carriers with an interval of 62.5 Hz between 62.5 Hz and 1500 Hz are selected as effective carriers. When the previous code rate requirement is less than 2.4Kbps, less carriers need to be used for modulation transmission.
b)载波选择好之后。将前面输入的串行码流进行串并转换,然后QPSK调制到各有效子载波上,然后做合并处理就得到调制的信号波形。b) After the carrier is selected. Perform serial-to-parallel conversion on the previously input serial code stream, then QPSK modulate it onto each effective sub-carrier, and then perform combination processing to obtain the modulated signal waveform.
c)接收端,将接收下来的调制信号在对应调制的每一个子载波上做解调处理。c) The receiving end performs demodulation processing on the received modulated signal on each subcarrier corresponding to the modulation.
d)解调出来的码流进行并串转换,得到与正交多载波调制之前的码流信号。d) Parallel-to-serial conversion is performed on the demodulated code stream to obtain a code stream signal before orthogonal multi-carrier modulation.
当采用基于OFDM的方法进行调制解调处理时,如图4所示,需要进行的操作步骤如下:When the OFDM-based method is used for modulation and demodulation processing, as shown in Figure 4, the steps to be performed are as follows:
在发送端:On the sender side:
a)首先,像上面有载波生成器正交多载波调制模式一样,首先需要选择子载波的数量和频率。根据频率间隔为62.5Hz。根据前面压缩码流的需求,按照QPSK调制的方式选择有效子载波的数量和频率,同样假设进来的码流为2.4Kbps,采用QPSK调制时,选择的正交子载波为62.5Hz到1500Hz之间间隔为62.5Hz的子载波。a) First, as with the carrier generator quadrature multi-carrier modulation mode above, the number and frequency of subcarriers need to be selected first. According to the frequency interval is 62.5Hz. According to the requirements of the previous compressed code stream, select the number and frequency of effective subcarriers according to the QPSK modulation method. Also assume that the incoming code stream is 2.4Kbps. When using QPSK modulation, the selected orthogonal subcarriers are between 62.5Hz and 1500Hz. Subcarriers spaced at 62.5Hz.
b)经过低码率压缩模块压缩后的小于2.4Kbps的码流首先经过串并转换,形成并行的码流。b) The code stream less than 2.4Kbps compressed by the low bit rate compression module is first converted into a parallel code stream through serial-to-parallel conversion.
c)将并行后的码流做QPSK调制,然后对应到128点IFFT处理前的向量值上。为保证复数IFFT运行结果是实数,128点IFFT前的向量点中,第1点和第65点取0值,压缩码流经QPSK调制后填充到第2点到第25点上,第26到64点填充0值,第128点填充0值。第2到64点和第66到128点关于第65点共轭偶对称。对填充好的向量进行IFFT变换,得到128个点的OFDM调制向量数据。c) Perform QPSK modulation on the parallel code stream, and then correspond to the vector value before 128-point IFFT processing. In order to ensure that the complex IFFT operation result is a real number, among the vector points before the 128-point IFFT, the 1st point and the 65th point take the value of 0, and the compressed code stream is filled to the 2nd to the 25th point after QPSK modulation, and the 26th to the 25th point. The 64th point is filled with 0 value, and the 128th point is filled with 0 value. The 2nd to 64th points and the 66th to 128th points are conjugate-even symmetrical about the 65th point. Perform IFFT transformation on the filled vector to obtain 128 points of OFDM modulation vector data.
d)这128点的数据,按照8KHz采样的过程其实是占16ms。经过同步模块,加上前面描述的4ms的同步序列,就可以输出到RPE-LTP声码器模块,接着后面的处理进行无线发射了。d) The data of these 128 points, according to the 8KHz sampling process, actually takes up 16ms. After the synchronization module, plus the 4ms synchronization sequence described above, it can be output to the RPE-LTP vocoder module, and then the subsequent processing is carried out for wireless transmission.
在接收端,做OFDM解调就可以解调出来压缩后的码流来。在做OFDM解调之前,需要首先经过同步模块做同步相关处理,找出同步点,然后从20ms的语音帧里面提取出调制数据的16ms信息来,进行OFDM解调处理。At the receiving end, OFDM demodulation can demodulate the compressed code stream. Before doing OFDM demodulation, it is necessary to do synchronization related processing through the synchronization module to find out the synchronization point, and then extract the 16ms information of the modulated data from the 20ms voice frame for OFDM demodulation processing.
最后,本发明形成的加密模块和解密模块可嵌入到手机内来实现端到端加密传输,也可以按照这个方法实现定制端到端加密手机。Finally, the encryption module and decryption module formed by the present invention can be embedded in the mobile phone to realize end-to-end encrypted transmission, and the end-to-end encrypted mobile phone can also be customized according to this method.
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