从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器

我会带着你远行 2021-11-01 20:42 430阅读 0赞

**从零开始写一个RTSP服务器系列**

[★我的开源项目-RtspServer][-RtspServer]

[从零开始写一个RTSP服务器（一）RTSP协议讲解][RTSP_RTSP]

[从零开始写一个RTSP服务器（二）RTSP协议的实现][RTSP_RTSP 1]

[从零开始写一个RTSP服务器（三）RTP传输H.264][RTSP_RTP_H.264]

[从零开始写一个RTSP服务器（四）一个传输H.264的RTSP服务器][RTSP_H.264_RTSP]

[从零开始写一个RTSP服务器（五）RTP传输AAC][RTSP_RTP_AAC]

[从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器][RTSP_AAC_RTSP]

[从零开始写一个RTSP服务器（七）多播传输RTP包][RTSP_RTP]

[从零开始写一个RTSP服务器（八）一个多播的RTSP服务器][RTSP_RTSP 2]

[从零开始写一个RTSP服务器（九）一个RTP OVER RTSP/TCP的RTSP服务器][RTSP_RTP OVER RTSP_TCP_RTSP]

# 从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器 #

### 文章目录 ###

*  从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器
 *   *  一、建立套接字
     *  二、接收客户端连接
     *  三、解析命令
     *  四、处理请求
     *   *  4.1 OPTIONS
         *  4.2 DESCRIBE
         *  4.3 SETUP
         *  4.4 PLAY
     *  五、AAC RTP打包发送
     *  六、源码
     *   *  aac\_rtsp\_server.c
         *  rtp.h
         *  rtp.c
     *  七、测试

本文主要是结合

[从零开始写一个RTSP服务器（一）不一样的RTSP协议讲解][RTSP_RTSP]

[从零开始写一个RTSP服务器（二）RTSP协议的实现][RTSP_RTSP 1]

[从零开始写一个RTSP服务器（五）RTP传输AAC][RTSP_RTP_AAC]

这几篇文章的内容总结，然后写出完成一个传输AAC的RTSP服务器，所有的知识点都在这几篇文章中，在看此篇文章前建议先认真看一看前面那几篇文章

本文不会讲解新知识点，如果你仔细看前几篇文章，相信你不需要看本文就能够完成一个传输AAC的RTSP服务器

下面主要是介绍我提供的示例代码的运行流程和其中的一些细节

## 一、建立套接字 ##

一开始进入main函数后，就监听服务器tcp套接字，绑定端口号，然后开始监听

然后再分别建立用于RTP和RTCP的udp套接字，绑定好端口

然后进入循环中开始服务

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    main()
    {
        
    	/* 创建服务器tcp套接字，绑定端口，监听 */
        serverSockfd = createTcpSocket();
    	bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
    	listen(serverSockfd, 10);
    	
        /* 建立用于RTP和RTCP的udp套接字，绑定好端口 */
        serverRtpSockfd = createUdpSocket();
        serverRtcpSockfd = createUdpSocket();
        bindSocketAddr(serverRtpSockfd, "0.0.0.0", SERVER_RTP_PORT);
        bindSocketAddr(serverRtcpSockfd, "0.0.0.0", SERVER_RTCP_PORT);
        
    	while(1)
    	{
        
            ...
    	}
    }

## 二、接收客户端连接 ##

在while循环中接收客户端，然后调用doClient服务

main()
    {
        
        ...
        while(1)
    	{
        
            clientSockfd = acceptClient(serverSockfd, clientIp, &clientPort);
            doClient(clientSockfd, clientIp, clientPort, serverRtpSockfd, serverRtcpSockfd);
    	}
    }

上面其实就是一个TCP服务器的基本步骤，没有什么特别的

下面来看一看doClient函数

## 三、解析命令 ##

doClient就是一个while循环（这是一个同时只能服务一个客户的服务器），不断地接收命令解析命令，然后调用相应地操作

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    doClient()
    {
        
    	while(1)
    	{
        
         	recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);   
            ...
            sscanf(line, "%s %s %s\r\n", method, url, version);
            ...
            sscanf(line, "CSeq: %d\r\n", &cseq)
            ...
    	}
    }

## 四、处理请求 ##

在解析完客户端命令后，会调用相应的请求，处理完之后讲接收打印到`sBuf`中，然后发送给客户端

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    doClient()
    {
        
    	while(1)
    	{
        
    		...
             /* 处理请求 */
             if(!strcmp(method, "OPTIONS"))
                 handleCmd_OPTIONS(sBuf, cseq);
            else if(!strcmp(method, "DESCRIBE"))
                handleCmd_DESCRIBE(sBuf, cseq, url);
            else if(!strcmp(method, "SETUP"))
                handleCmd_SETUP(sBuf, cseq, clientRtpPort);
            else if(!strcmp(method, "PLAY"))
                handleCmd_PLAY(sBuf, cseq);
            
            /* 放回结果 */
            send(clientSockfd, sBuf, strlen(sBuf), 0);
    	}
    }

下面来看看各个请求的行动

### 4.1 OPTIONS ###

返回可用方法

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    static int handleCmd_OPTIONS(char* result, int cseq)
    {
        
        sprintf(result, "RTSP/1.0 200 OK\r\n"
                        "CSeq: %d\r\n"
                        "Public: OPTIONS, DESCRIBE, SETUP, PLAY\r\n"
                        "\r\n",
                        cseq);
                    
        return 0;
    }

### 4.2 DESCRIBE ###

返回sdp文件信息，注意这个示例的sdp文件和[从零开始写一个RTSP服务器（四）一个传输H.264的RTSP服务器][RTSP_H.264_RTSP]中的sdp文件是不一样的，这是很重要的

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    static int handleCmd_DESCRIBE(char* result, int cseq, char* url)
    {
        
        char sdp[500];
        char localIp[100];
    
        sscanf(url, "rtsp://%[^:]:", localIp);
    
        sprintf(sdp, "v=0\r\n"
                     "o=- 9%ld 1 IN IP4 %s\r\n"
                     "t=0 0\r\n"
                     "a=control:*\r\n"
                     "m=audio 0 RTP/AVP 97\r\n"
                     "a=rtpmap:97 mpeg4-generic/44100/2\r\n"
                     "a=fmtp:97 SizeLength=13;\r\n"
                     "a=control:track0\r\n",
                     time(NULL), localIp);
        
        sprintf(result, "RTSP/1.0 200 OK\r\nCSeq: %d\r\n"
                        "Content-Base: %s\r\n"
                        "Content-type: application/sdp\r\n"
                        "Content-length: %d\r\n\r\n"
                        "%s",
                        cseq,
                        url,
                        strlen(sdp),
                        sdp);
        
        return 0;
    }

### 4.3 SETUP ###

SETUP过程发送服务端RTP端口和RTCP端口

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    static int handleCmd_SETUP(char* result, int cseq, int clientRtpPort)
    {
        
        sprintf(result, "RTSP/1.0 200 OK\r\n"
                        "CSeq: %d\r\n"
                        "Transport: RTP/AVP;unicast;client_port=%d-%d;server_port=%d-%d\r\n"
                        "Session: 66334873\r\n"
                        "\r\n",
                        cseq,
                        clientRtpPort,
                        clientRtpPort+1,
                        SERVER_RTP_PORT,
                        SERVER_RTCP_PORT);
        
        return 0;
    }

### 4.4 PLAY ###

PLAY操作回复后，会开始发送RTP包

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    static int handleCmd_PLAY(char* result, int cseq)
    {
        
        sprintf(result, "RTSP/1.0 200 OK\r\n"
                        "CSeq: %d\r\n"
                        "Range: npt=0.000-\r\n"
                        "Session: 66334873; timeout=60\r\n\r\n",
                        cseq);
        
        return 0;
    }

## 五、AAC RTP打包发送 ##

先读取ADTS头，得到一帧的大小，然后再读取AAC Data，再通过RTP打包传输

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    doClient()
    {
        
    	while(1)
    	{
        
    		...
    
             send(clientSockfd, sBuf, strlen(sBuf), 0);
            
    		if(!strcmp(method, "PLAY"))
    		{
        
                while(1)
                {
        
                    /* 读取ADTS头部 */
                	read(fd, frame, 7);
                    
                    /* 解析头部 */
                    parseAdtsHeader(frame, &adtsHeader);
                    
                    /* 读取一帧 */
                    read(fd, frame, adtsHeader.aacFrameLength-7);
                    
                    /* RTP打包发送 */
                    rtpSendAACFrame(localRtpSockfd, clientIP, clientRtpPort,
                                    rtpPacket, frame, adtsHeader.aacFrameLength-7);
    		
                }
            }
    
        }
    }

看一看AAC的RTP打包发送过程

先填充RTP载荷前4个字节，然后发送RTP包

发送后序列号增加，时间戳增加

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    static int rtpSendAACFrame(int socket, const char* ip, int16_t port,
                                struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize)
    {
        
        /* 填充前4个字节 */
        rtpPacket->payload[0] = 0x00;
        rtpPacket->payload[1] = 0x10;
        rtpPacket->payload[2] = (frameSize & 0x1FE0) >> 5; //高8位
        rtpPacket->payload[3] = (frameSize & 0x1F) << 3; //低5位
        
        /* 发送RTP包 */
        memcpy(rtpPacket->payload+4, frame, frameSize);
        rtpSendPacket(socket, ip, port, rtpPacket, frameSize+4);
        
        /* 增加序列号和时间戳 */
        rtpPacket->rtpHeader.seq++;
        rtpPacket->rtpHeader.timestamp += 1025;
    }

## 六、源码 ##

总共由三个文件`aac_rtsp_server.c`、`rtp.h`、`rtp.c`

### aac\_rtsp\_server.c ###

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    #include <stdio.h>
    #include <stdlib.h>
    #include <stdint.h>
    #include <string.h>
    #include <sys/types.h>
    #include <sys/socket.h>
    #include <sys/socket.h>
    #include <netinet/in.h>
    #include <arpa/inet.h>
    #include <time.h>
    #include <sys/types.h>
    #include <sys/stat.h>
    #include <fcntl.h>
    
    #include "rtp.h"
    
    #define SERVER_PORT     8554
    #define SERVER_RTP_PORT  55532
    #define SERVER_RTCP_PORT 55533
    #define BUF_MAX_SIZE    (1024*1024)
    #define AAC_FILE_NAME   "test.aac"
    
    static int createTcpSocket()
    {
        
        int sockfd;
        int on = 1;
    
        sockfd = socket(AF_INET, SOCK_STREAM, 0);
        if(sockfd < 0)
            return -1;
    
        setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on));
    
        return sockfd;
    }
    
    static int createUdpSocket()
    {
        
        int sockfd;
        int on = 1;
    
        sockfd = socket(AF_INET, SOCK_DGRAM, 0);
        if(sockfd < 0)
            return -1;
    
        setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on));
    
        return sockfd;
    }
    
    static int bindSocketAddr(int sockfd, const char* ip, int port)
    {
        
        struct sockaddr_in addr;
    
        addr.sin_family = AF_INET;
        addr.sin_port = htons(port);
        addr.sin_addr.s_addr = inet_addr(ip);
    
        if(bind(sockfd, (struct sockaddr *)&addr, sizeof(struct sockaddr)) < 0)
            return -1;
    
        return 0;
    }
    
    struct AdtsHeader
    {
        
        unsigned int syncword;  //12 bit 同步字 '1111 1111 1111'，说明一个ADTS帧的开始
        unsigned int id;        //1 bit MPEG 标示符， 0 for MPEG-4，1 for MPEG-2
        unsigned int layer;     //2 bit 总是'00'
        unsigned int protectionAbsent;  //1 bit 1表示没有crc，0表示有crc
        unsigned int profile;           //1 bit 表示使用哪个级别的AAC
        unsigned int samplingFreqIndex; //4 bit 表示使用的采样频率
        unsigned int privateBit;        //1 bit
        unsigned int channelCfg; //3 bit 表示声道数
        unsigned int originalCopy;         //1 bit 
        unsigned int home;                  //1 bit 
    
        /*下面的为改变的参数即每一帧都不同*/
        unsigned int copyrightIdentificationBit;   //1 bit
        unsigned int copyrightIdentificationStart; //1 bit
        unsigned int aacFrameLength;               //13 bit 一个ADTS帧的长度包括ADTS头和AAC原始流
        unsigned int adtsBufferFullness;           //11 bit 0x7FF 说明是码率可变的码流
    
        /* number_of_raw_data_blocks_in_frame
         * 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧
         * 所以说number_of_raw_data_blocks_in_frame == 0 
         * 表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据)
         */
        unsigned int numberOfRawDataBlockInFrame; //2 bit
    };
    
    static int parseAdtsHeader(uint8_t* in, struct AdtsHeader* res)
    {
        
        static int frame_number = 0;
        memset(res,0,sizeof(*res));
    
        if ((in[0] == 0xFF)&&((in[1] & 0xF0) == 0xF0))
        {
        
            res->id = ((unsigned int) in[1] & 0x08) >> 3;
            res->layer = ((unsigned int) in[1] & 0x06) >> 1;
            res->protectionAbsent = (unsigned int) in[1] & 0x01;
            res->profile = ((unsigned int) in[2] & 0xc0) >> 6;
            res->samplingFreqIndex = ((unsigned int) in[2] & 0x3c) >> 2;
            res->privateBit = ((unsigned int) in[2] & 0x02) >> 1;
            res->channelCfg = ((((unsigned int) in[2] & 0x01) << 2) | (((unsigned int) in[3] & 0xc0) >> 6));
            res->originalCopy = ((unsigned int) in[3] & 0x20) >> 5;
            res->home = ((unsigned int) in[3] & 0x10) >> 4;
            res->copyrightIdentificationBit = ((unsigned int) in[3] & 0x08) >> 3;
            res->copyrightIdentificationStart = (unsigned int) in[3] & 0x04 >> 2;
            res->aacFrameLength = (((((unsigned int) in[3]) & 0x03) << 11) |
                                    (((unsigned int)in[4] & 0xFF) << 3) |
                                        ((unsigned int)in[5] & 0xE0) >> 5) ;
            res->adtsBufferFullness = (((unsigned int) in[5] & 0x1f) << 6 |
                                            ((unsigned int) in[6] & 0xfc) >> 2);
            res->numberOfRawDataBlockInFrame = ((unsigned int) in[6] & 0x03);
    
            return 0;
        }
        else
        {
        
            printf("failed to parse adts header\n");
            return -1;
        }
    }
    
    static int rtpSendAACFrame(int socket, const char* ip, int16_t port,
                                struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize)
    {
        
        int ret;
    
        rtpPacket->payload[0] = 0x00;
        rtpPacket->payload[1] = 0x10;
        rtpPacket->payload[2] = (frameSize & 0x1FE0) >> 5; //高8位
        rtpPacket->payload[3] = (frameSize & 0x1F) << 3; //低5位
    
        memcpy(rtpPacket->payload+4, frame, frameSize);
    
        ret = rtpSendPacket(socket, ip, port, rtpPacket, frameSize+4);
        if(ret < 0)
        {
        
            printf("failed to send rtp packet\n");
            return -1;
        }
    
        rtpPacket->rtpHeader.seq++;
    
        /*
         * 如果采样频率是44100
         * 一般AAC每个1024个采样为一帧
         * 所以一秒就有 44100 / 1024 = 43帧
         * 时间增量就是 44100 / 43 = 1025
         * 一帧的时间为 1 / 43 = 23ms
         */
        rtpPacket->rtpHeader.timestamp += 1025;
    
        return 0;
    }
    
    static int acceptClient(int sockfd, char* ip, int* port)
    {
        
        int clientfd;
        socklen_t len = 0;
        struct sockaddr_in addr;
    
        memset(&addr, 0, sizeof(addr));
        len = sizeof(addr);
    
        clientfd = accept(sockfd, (struct sockaddr *)&addr, &len);
        if(clientfd < 0)
            return -1;
        
        strcpy(ip, inet_ntoa(addr.sin_addr));
        *port = ntohs(addr.sin_port);
    
        return clientfd;
    }
    
    static char* getLineFromBuf(char* buf, char* line)
    {
        
        while(*buf != '\n')
        {
        
            *line = *buf;
            line++;
            buf++;
        }
    
        *line = '\n';
        ++line;
        *line = '\0';
    
        ++buf;
        return buf; 
    }
    
    static int handleCmd_OPTIONS(char* result, int cseq)
    {
        
        sprintf(result, "RTSP/1.0 200 OK\r\n"
                        "CSeq: %d\r\n"
                        "Public: OPTIONS, DESCRIBE, SETUP, PLAY\r\n"
                        "\r\n",
                        cseq);
                    
        return 0;
    }
    
    static int handleCmd_DESCRIBE(char* result, int cseq, char* url)
    {
        
        char sdp[500];
        char localIp[100];
    
        sscanf(url, "rtsp://%[^:]:", localIp);
    
        sprintf(sdp, "v=0\r\n"
                     "o=- 9%ld 1 IN IP4 %s\r\n"
                     "t=0 0\r\n"
                     "a=control:*\r\n"
                     "m=audio 0 RTP/AVP 97\r\n"
                     "a=rtpmap:97 mpeg4-generic/44100/2\r\n"
                     "a=fmtp:97 SizeLength=13;\r\n"
                     "a=control:track0\r\n",
                     time(NULL), localIp);
    
        sprintf(result, "RTSP/1.0 200 OK\r\nCSeq: %d\r\n"
                        "Content-Base: %s\r\n"
                        "Content-type: application/sdp\r\n"
                        "Content-length: %d\r\n\r\n"
                        "%s",
                        cseq,
                        url,
                        strlen(sdp),
                        sdp);
        
        return 0;
    }
    
    static int handleCmd_SETUP(char* result, int cseq, int clientRtpPort)
    {
        
        sprintf(result, "RTSP/1.0 200 OK\r\n"
                        "CSeq: %d\r\n"
                        "Transport: RTP/AVP;unicast;client_port=%d-%d;server_port=%d-%d\r\n"
                        "Session: 66334873\r\n"
                        "\r\n",
                        cseq,
                        clientRtpPort,
                        clientRtpPort+1,
                        SERVER_RTP_PORT,
                        SERVER_RTCP_PORT
                        );
        
        return 0;
    }
    
    static int handleCmd_PLAY(char* result, int cseq)
    {
        
        sprintf(result, "RTSP/1.0 200 OK\r\n"
                        "CSeq: %d\r\n"
                        "Range: npt=0.000-\r\n"
                        "Session: 66334873; timeout=60\r\n\r\n",
                        cseq);
        
        return 0;
    }
    
    static void doClient(int clientSockfd, const char* clientIP, int clientPort,
                            int serverRtpSockfd, int serverRtcppSockfd)
    {
        
        char method[40];
        char url[100];
        char version[40];
        int cseq;
        int clientRtpPort, clientRtcpPort;
        char *bufPtr;
        char* rBuf = malloc(BUF_MAX_SIZE);
        char* sBuf = malloc(BUF_MAX_SIZE);
        char line[400];
    
        while(1)
        {
        
            int recvLen;
    
            recvLen = recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
            if(recvLen <= 0)
                goto out;
    
            rBuf[recvLen] = '\0';
            printf("---------------C->S--------------\n");
            printf("%s", rBuf);
    
            /* 解析方法 */
            bufPtr = getLineFromBuf(rBuf, line);
            if(sscanf(line, "%s %s %s\r\n", method, url, version) != 3)
            {
        
                printf("parse err\n");
                goto out;
            }
    
            /* 解析序列号 */
            bufPtr = getLineFromBuf(bufPtr, line);
            if(sscanf(line, "CSeq: %d\r\n", &cseq) != 1)
            {
        
                printf("parse err\n");
                goto out;
            }
    
            /* 如果是SETUP，那么就再解析client_port */
            if(!strcmp(method, "SETUP"))
            {
        
                while(1)
                {
        
                    bufPtr = getLineFromBuf(bufPtr, line);
                    if(!strncmp(line, "Transport:", strlen("Transport:")))
                    {
        
                        sscanf(line, "Transport: RTP/AVP;unicast;client_port=%d-%d\r\n",
                                        &clientRtpPort, &clientRtcpPort);
                        break;
                    }
                }
            }
    
            if(!strcmp(method, "OPTIONS"))
            {
        
                if(handleCmd_OPTIONS(sBuf, cseq))
                {
        
                    printf("failed to handle options\n");
                    goto out;
                }
            }
            else if(!strcmp(method, "DESCRIBE"))
            {
        
                if(handleCmd_DESCRIBE(sBuf, cseq, url))
                {
        
                    printf("failed to handle describe\n");
                    goto out;
                }
            }
            else if(!strcmp(method, "SETUP"))
            {
        
                if(handleCmd_SETUP(sBuf, cseq, clientRtpPort))
                {
        
                    printf("failed to handle setup\n");
                    goto out;
                }
            }
            else if(!strcmp(method, "PLAY"))
            {
        
                if(handleCmd_PLAY(sBuf, cseq))
                {
        
                    printf("failed to handle play\n");
                    goto out;
                }
            }
            else
            {
        
                goto out;
            }
    
            printf("---------------S->C--------------\n");
            printf("%s", sBuf);
            send(clientSockfd, sBuf, strlen(sBuf), 0);
    
            if(!strcmp(method, "PLAY"))
            {
        
                struct AdtsHeader adtsHeader;
                struct RtpPacket* rtpPacket;
                uint8_t* frame;
                int ret;
    
                int fd = open(AAC_FILE_NAME, O_RDONLY);
                if(fd < 0)
                {
        
                    printf("failed to open %s\n", AAC_FILE_NAME);
                    goto out;
                }
    
                frame = (uint8_t*)malloc(5000);
                rtpPacket = malloc(5000);
    
                rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_AAC, 1, 0, 0, 0x32411);
    
                while(1)
                {
        
                    ret = read(fd, frame, 7);
                    if(ret <= 0)
                    {
        
                        break;
                    }
    
                    if(parseAdtsHeader(frame, &adtsHeader) < 0)
                    {
        
                        printf("parse err\n");
                        break;
                    }
    
                    ret = read(fd, frame, adtsHeader.aacFrameLength-7);
                    if(ret < 0)
                    {
        
                        printf("read err\n");
                        break;
                    }
    
                    rtpSendAACFrame(serverRtpSockfd, clientIP, clientRtpPort,
                                    rtpPacket, frame, adtsHeader.aacFrameLength-7);
    
                    usleep(23000);
                }
    
                free(frame);
                free(rtpPacket);
            }
        }
    out:
        close(clientSockfd);
        free(rBuf);
        free(sBuf);
    }
    
    int main(int argc, char* argv[])
    {
        
        int serverSockfd;
        int serverRtpSockfd, serverRtcpSockfd;
        int ret;
    
        serverSockfd = createTcpSocket();
        if(serverSockfd < 0)
        {
        
            printf("failed to create tcp socket\n");
            return -1;
        }
    
        ret = bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
        if(ret < 0)
        {
        
            printf("failed to bind addr\n");
            return -1;
        }
    
        ret = listen(serverSockfd, 10);
        if(ret < 0)
        {
        
            printf("failed to listen\n");
            return -1;
        }
    
        serverRtpSockfd = createUdpSocket();
        serverRtcpSockfd = createUdpSocket();
        if(serverRtpSockfd < 0 || serverRtcpSockfd < 0)
        {
        
            printf("failed to create udp socket\n");
            return -1;
        }
    
        if(bindSocketAddr(serverRtpSockfd, "0.0.0.0", SERVER_RTP_PORT) < 0 ||
            bindSocketAddr(serverRtcpSockfd, "0.0.0.0", SERVER_RTCP_PORT) < 0)
        {
        
            printf("failed to bind addr\n");
            return -1;
        }
    
        printf("rtsp://127.0.0.1:%d\n", SERVER_PORT);
    
        while(1)
        {
        
            int clientSockfd;
            char clientIp[40];
            int clientPort;
    
            clientSockfd = acceptClient(serverSockfd, clientIp, &clientPort);
            if(clientSockfd < 0)
            {
        
                printf("failed to accept client\n");
                return -1;
            }
    
            printf("accept client;client ip:%s,client port:%d\n", clientIp, clientPort);
    
            doClient(clientSockfd, clientIp, clientPort, serverRtpSockfd, serverRtcpSockfd);
        }
    
        return 0;
    }

### rtp.h ###

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    #ifndef _RTP_H_
    #define _RTP_H_
    #include <stdint.h>
    
    #define RTP_VESION              2
    
    #define RTP_PAYLOAD_TYPE_H264   96
    #define RTP_PAYLOAD_TYPE_AAC    97
    
    #define RTP_HEADER_SIZE         12
    #define RTP_MAX_PKT_SIZE        1400
    
    /*
     *
     *    0                   1                   2                   3
     *    7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0
     *   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     *   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
     *   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     *   |                           timestamp                           |
     *   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     *   |           synchronization source (SSRC) identifier            |
     *   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
     *   |            contributing source (CSRC) identifiers             |
     *   :                             ....                              :
     *   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     *
     */
    struct RtpHeader
    {
        
        /* byte 0 */
        uint8_t csrcLen:4;
        uint8_t extension:1;
        uint8_t padding:1;
        uint8_t version:2;
    
        /* byte 1 */
        uint8_t payloadType:7;
        uint8_t marker:1;
        
        /* bytes 2,3 */
        uint16_t seq;
        
        /* bytes 4-7 */
        uint32_t timestamp;
        
        /* bytes 8-11 */
        uint32_t ssrc;
    };
    
    struct RtpPacket
    {
        
        struct RtpHeader rtpHeader;
        uint8_t payload[0];
    };
    
    void rtpHeaderInit(struct RtpPacket* rtpPacket, uint8_t csrcLen, uint8_t extension,
                        uint8_t padding, uint8_t version, uint8_t payloadType, uint8_t marker,
                       uint16_t seq, uint32_t timestamp, uint32_t ssrc);
    int rtpSendPacket(int socket, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint32_t dataSize);
    
    #endif //_RTP_H_

### rtp.c ###

/*
     * 作者：_JT_
     * 博客：https://blog.csdn.net/weixin_42462202
     */
    
    #include <sys/types.h>
    #include <sys/socket.h>
    #include <arpa/inet.h>
    #include <netinet/in.h>
    #include <arpa/inet.h>
    
    #include "rtp.h"
    
    void rtpHeaderInit(struct RtpPacket* rtpPacket, uint8_t csrcLen, uint8_t extension,
                        uint8_t padding, uint8_t version, uint8_t payloadType, uint8_t marker,
                       uint16_t seq, uint32_t timestamp, uint32_t ssrc)
    {
        
        rtpPacket->rtpHeader.csrcLen = csrcLen;
        rtpPacket->rtpHeader.extension = extension;
        rtpPacket->rtpHeader.padding = padding;
        rtpPacket->rtpHeader.version = version;
        rtpPacket->rtpHeader.payloadType =  payloadType;
        rtpPacket->rtpHeader.marker = marker;
        rtpPacket->rtpHeader.seq = seq;
        rtpPacket->rtpHeader.timestamp = timestamp;
        rtpPacket->rtpHeader.ssrc = ssrc;
    }
    
    int rtpSendPacket(int socket, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint32_t dataSize)
    {
        
        struct sockaddr_in addr;
        int ret;
    
        addr.sin_family = AF_INET;
        addr.sin_port = htons(port);
        addr.sin_addr.s_addr = inet_addr(ip);
    
        rtpPacket->rtpHeader.seq = htons(rtpPacket->rtpHeader.seq);
        rtpPacket->rtpHeader.timestamp = htonl(rtpPacket->rtpHeader.timestamp);
        rtpPacket->rtpHeader.ssrc = htonl(rtpPacket->rtpHeader.ssrc);
    
        ret = sendto(socket, (void*)rtpPacket, dataSize+RTP_HEADER_SIZE, 0,
                        (struct sockaddr*)&addr, sizeof(addr));
    
        rtpPacket->rtpHeader.seq = ntohs(rtpPacket->rtpHeader.seq);
        rtpPacket->rtpHeader.timestamp = ntohl(rtpPacket->rtpHeader.timestamp);
        rtpPacket->rtpHeader.ssrc = ntohl(rtpPacket->rtpHeader.ssrc);
    
        return ret;
    }

## 七、测试 ##

将`aac_rtsp_server.c`、`rtp.h`、`rtp.c`这三个文件保存下来

编译运行，默认会打开`test.aac`的音频文件，如果你没有音频源的话，可以从[RtspServer][]的example目录下获取

# gcc aac_rtsp_server.c rtp.c
    # ./a.out

运行之后会打印一个url

rtsp://127.0.0.1:8554

再vlc打开即可听到音频

[-RtspServer]: https://blog.csdn.net/weixin_42462202/article/details/98956346
[RTSP_RTSP]: https://blog.csdn.net/weixin_42462202/article/details/98986535
[RTSP_RTSP 1]: https://blog.csdn.net/weixin_42462202/article/details/99068041
[RTSP_RTP_H.264]: https://blog.csdn.net/weixin_42462202/article/details/99089020
[RTSP_H.264_RTSP]: https://blog.csdn.net/weixin_42462202/article/details/99111635
[RTSP_RTP_AAC]: https://blog.csdn.net/weixin_42462202/article/details/99200935
[RTSP_AAC_RTSP]: https://blog.csdn.net/weixin_42462202/article/details/99311193
[RTSP_RTP]: https://blog.csdn.net/weixin_42462202/article/details/99329048
[RTSP_RTSP 2]: https://blog.csdn.net/weixin_42462202/article/details/99413781
[RTSP_RTP OVER RTSP_TCP_RTSP]: https://blog.csdn.net/weixin_42462202/article/details/99442338
[RtspServer]: https://github.com/ImSjt/RtspServer