【GB28181】基于RTSP协议的简易流媒体服务器

概述

学习的demo来自于B站“北小菜”博主。主要基于该demo理解RTSP服务器的框架。该demo主要实现了从本地H264视频文件中读取视频帧，然后通过RTP协议实时传输给RTSP客户端

主要逻辑

• 初始化，创建RTSP服务器监听套接字，绑定地址和端口号，然后开始监听
• 主循环
  • 不断的接收客户端连接然后进行处理
  • 打印服务器信息，并不对其进行处理

RTSP客户端

static void doClient(int clientSockfd, const char* clientIP, int clientPort) {

    char method[40];
    char url[100];
    char version[40];
    int CSeq;

    int serverRtpSockfd = -1, serverRtcpSockfd = -1;
    int clientRtpPort, clientRtcpPort;
    char* rBuf = (char*)malloc(BUF_MAX_SIZE);
    char* sBuf = (char*)malloc(BUF_MAX_SIZE);

    while (true) {
        int recvLen;

        recvLen = recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
        if (recvLen <= 0) {
            break;
        }

        rBuf[recvLen] = '\0';
        printf("%s rBuf = %s \n",__FUNCTION__,rBuf);

        const char* sep = "\n";
        char* line = strtok(rBuf, sep);
        while (line) {
            if (strstr(line, "OPTIONS") ||
                strstr(line, "DESCRIBE") ||
                strstr(line, "SETUP") ||
                strstr(line, "PLAY")) {

                if (sscanf(line, "%s %s %s\r\n", method, url, version) != 3) {
                    // error
                }
            }
            else if (strstr(line, "CSeq")) {
                if (sscanf(line, "CSeq: %d\r\n", &CSeq) != 1) {
                    // error
                }
            }
            else if (!strncmp(line, "Transport:", strlen("Transport:"))) {
                // Transport: RTP/AVP/UDP;unicast;client_port=13358-13359
                // Transport: RTP/AVP;unicast;client_port=13358-13359

                if (sscanf(line, "Transport: RTP/AVP/UDP;unicast;client_port=%d-%d\r\n",
                    &clientRtpPort, &clientRtcpPort) != 2) {
                    // error
                    printf("parse Transport error \n");
                }
            }
            line = strtok(NULL, sep);
        }

        if (!strcmp(method, "OPTIONS")) {
            if (handleCmd_OPTIONS(sBuf, CSeq))
            {
                printf("failed to handle options\n");
                break;
            }
        }
        else if (!strcmp(method, "DESCRIBE")) {
            if (handleCmd_DESCRIBE(sBuf, CSeq, url))
            {
                printf("failed to handle describe\n");
                break;
            }
        }
        else if (!strcmp(method, "SETUP")) {
            if (handleCmd_SETUP(sBuf, CSeq, clientRtpPort))
            {
                printf("failed to handle setup\n");
                break;
            }

            serverRtpSockfd = createUdpSocket();
            serverRtcpSockfd = createUdpSocket();

            if (serverRtpSockfd < 0 || serverRtcpSockfd < 0)
            {
                printf("failed to create udp socket\n");
                break;
            }

            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");
                break;
            }

        }
        else if (!strcmp(method, "PLAY")) {
            if (handleCmd_PLAY(sBuf, CSeq))
            {
                printf("failed to handle play\n");
                break;
            }
        }
        else {
            printf("未定义的method = %s \n", method);
            break;
        }
        printf("sBuf = %s \n", sBuf);
        printf("%s sBuf = %s \n", __FUNCTION__, sBuf);

        send(clientSockfd, sBuf, strlen(sBuf), 0);


        //开始播放，发送RTP包
        if (!strcmp(method, "PLAY")) {

            int frameSize, startCode;
            char* frame = (char*)malloc(500000);
            struct RtpPacket* rtpPacket = (struct RtpPacket*)malloc(500000);
            FILE* fp = fopen(H264_FILE_NAME, "rb");
            if (!fp) {
                printf("读取 %s 失败\n", H264_FILE_NAME);
                break;
            }
            rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_H264, 0,
                0, 0, 0x88923423);

            printf("start play\n");
            printf("client ip:%s\n", clientIP);
            printf("client port:%d\n", clientRtpPort);

            while (true) {
                frameSize = getFrameFromH264File(fp, frame, 500000);
                if (frameSize < 0)
                {
                    printf("读取%s结束,frameSize=%d \n", H264_FILE_NAME, frameSize);
                    break;
                }

                if (startCode3(frame))
                    startCode = 3;
                else
                    startCode = 4;

                frameSize -= startCode;
                rtpSendH264Frame(serverRtpSockfd, clientIP, clientRtpPort,
                    rtpPacket, frame + startCode, frameSize);

               

                Sleep(40);
                //usleep(40000);//1000/25 * 1000
            }
            free(frame);
            free(rtpPacket);

            break;
        }

        memset(method,0,sizeof(method)/sizeof(char));
        memset(url,0,sizeof(url)/sizeof(char));
        CSeq = 0;

    }

    closesocket(clientSockfd);
    if (serverRtpSockfd) {
        closesocket(serverRtpSockfd);
    }
    if (serverRtcpSockfd > 0) {
        closesocket(serverRtcpSockfd);
    }

    free(rBuf);
    free(sBuf);

}

逻辑分析

首先接收并解析RTSP请求；获取RTSP请求，分析其命令；然后通过分支语句对不同的RTSP命令进行处理并最终生成响应

RTP流传输（代码事例中具体通过PLAY命令处理）

• 首先是打开H264文件
• 然后初始化RTP包头，循环读取H264帧并发送RTP包
• 代码中使用的是模拟帧率的方法，真实代码实现中需要根据实际情况设置

RTSP协议交互

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=video 0 RTP/AVP 96\r\n"
        "a=rtpmap:96 H264/90000\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: %zu\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=10\r\n\r\n",
        cseq);

    return 0;
}

主要流程

• OPTIONS: 客户端查询服务器支持的命令
• DESCRIBE: 客户端请求服务器描述媒体会话，服务器返回SDP (Session Descr