基于tcp的RTP传输服务器

基于TCP的RTP传输服务器

流程

1，客户端请求RTSP的Describe请求时，RTSP服务器返回的SDP协议

2，客户端请求RTSP的Setup请求时，RTSP服务器不需要再对应创建RTP和RTCP的UDP连接通道，因为TCP版的RTP传输，客户端与服务器交互时，无论是RTSP信令还是RTP数据包或者是RTCP数据包，都是使用同一个tcp连接通道。

不过这个tcp连接通道在发送rtp数据包或者rtcp数据包时，需要加一些分隔字节。

3，客户端请求RTSP的Play请求时，RTSP服务器在对Play请求回复以后，还需要源源不断的同时向客户端发送音频流和视频流的RTP数据包。

与UDP 对比

  • UDP协议上的RTSP/RTP需要打开许多UDP端口,一个端口用于RTSP通信，n个端口用于RTP，n个端口用于RTCP
  • 中间网络路由器很容易就过滤或者忽略掉UDP数据包
  • UDP是不可靠传输协议，媒体包在因特网上传输时会面临着丢包

不需要再去创建UDP通道

CODE

​ //“m=” 出现新流 96代表H264 // 97代表音频流AAC

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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/TCP 96\r\n"
    "a=rtpmap:96 H264/90000\r\n"
    "a=control:track0\r\n"
    "m=audio 1 RTP/AVP/TCP 97\r\n"
    "a=rtpmap:97 mpeg4-generic/44100/2\r\n"
    "a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1210;\r\n"
    "a=control:track1\r\n",

    time(NULL), localIp);

sdp 返回到客户端后，客户端对每一路流都发起setup 发起的时候使用这里定义的名字 track0

doClient Function

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std::thread t1([&]() {
    
    int frameSize, startCode;
    char* frame = (char*)malloc(500000);
    // 创建RTP Packet
    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);
        return;
    }
    //RTP_PAYLOAD_TYPE_H264 规定类型 0x88923423 SSRC指定
    rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_H264, 0,
        0, 0, 0x88923423);

    printf("start play\n");

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

        // 判断分隔符
        if (startCode3(frame))
            startCode = 3;
        else
            startCode = 4;

        frameSize -= startCode;
        rtpSendH264Frame(clientSockfd, rtpPacket, frame + startCode, frameSize);

        rtpPacket->rtpHeader.timestamp += 90000 / 25;
    
        //Sleep(40);//->30,20,需要处理NAL类型
        Sleep(20);
        //usleep(40000);//1000/25 * 1000
    }
    free(frame);
    free(rtpPacket);
    
});

RTP的发送

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static int rtpSendH264Frame(int clientSockfd,
    struct RtpPacket* rtpPacket, char* frame, uint32_t frameSize)
{

    uint8_t naluType; // nalu第一个字节
    int sendByte = 0;
    int ret;

    naluType = frame[0];
    
    printf("%s frameSize=%d \n", __FUNCTION__, frameSize);

    if (frameSize <= RTP_MAX_PKT_SIZE) // nalu长度小于最大包场：单一NALU单元模式
    {

         //*   0 1 2 3 4 5 6 7 8 9
         //*  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
         //*  |F|NRI|  Type   | a single NAL unit ... |
         //*  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        memcpy(rtpPacket->payload, frame, frameSize);
        // 主要区别 RTP 0X00
        ret = rtpSendPacketOverTcp(clientSockfd, rtpPacket, frameSize,0x00);
        if(ret < 0)
            return -1;

        rtpPacket->rtpHeader.seq++;
        sendByte += ret;
        if ((naluType & 0x1F) == 7 || (naluType & 0x1F) == 8) // 如果是SPS、PPS就不需要加时间戳
        {

        }

    }
    else // nalu长度小于最大包：分片模式
    {

         //*  0                   1                   2
         //*  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
         //* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
         //* | FU indicator  |   FU header   |   FU payload   ...  |
         //* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



         //*     FU Indicator
         //*    0 1 2 3 4 5 6 7
         //*   +-+-+-+-+-+-+-+-+
         //*   |F|NRI|  Type   |
         //*   +---------------+



         //*      FU Header
         //*    0 1 2 3 4 5 6 7
         //*   +-+-+-+-+-+-+-+-+
         //*   |S|E|R|  Type   |
         //*   +---------------+


        int pktNum = frameSize / RTP_MAX_PKT_SIZE;       // 有几个完整的包
        int remainPktSize = frameSize % RTP_MAX_PKT_SIZE; // 剩余不完整包的大小
        int i, pos = 1;

        // 发送完整的包
        for (i = 0; i < pktNum; i++)
        {
            rtpPacket->payload[0] = (naluType & 0x60) | 28;
            rtpPacket->payload[1] = naluType & 0x1F;

            if (i == 0) //第一包数据
                rtpPacket->payload[1] |= 0x80; // start
            else if (remainPktSize == 0 && i == pktNum - 1) //最后一包数据
                rtpPacket->payload[1] |= 0x40; // end

            memcpy(rtpPacket->payload+2, frame+pos, RTP_MAX_PKT_SIZE);
            ret = rtpSendPacketOverTcp(clientSockfd, rtpPacket, RTP_MAX_PKT_SIZE+2,0x00);
            if(ret < 0)
                return -1;

            rtpPacket->rtpHeader.seq++;
            sendByte += ret;
            pos += RTP_MAX_PKT_SIZE;
        }

        // 发送剩余的数据
        if (remainPktSize > 0)
        {
            rtpPacket->payload[0] = (naluType & 0x60) | 28;
            rtpPacket->payload[1] = naluType & 0x1F;
            rtpPacket->payload[1] |= 0x40; //end

            memcpy(rtpPacket->payload+2, frame+pos, remainPktSize+2);
            ret = rtpSendPacketOverTcp(clientSockfd, rtpPacket, remainPktSize+2, 0x00);
            if(ret < 0)
                return -1;

            rtpPacket->rtpHeader.seq++;
            sendByte += ret;
        }
    }


    return sendByte;

}

rtpSendPacketOverTcp fuction

在发送之前 需要在RTP发送的字节之前加入四个字节

视频流和音频流都有两个通道 RTP和RTCP

如果不是I帧P帧的某一帧 则不能累加时间戳 否则会导致音视频传输不对等 如果累加了时间戳 会导致视频延时

tempBuf[0] = 0x24;//$ 分隔符
tempBuf[1] = channel;// 0x00; channel 一路流两个通道
tempBuf[2] = (uint8_t)(((rtpSize) & 0xFF00) >> 8);
tempBuf[3] = (uint8_t)((rtpSize) & 0xFF);

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int rtpSendPacketOverTcp(int clientSockfd, struct RtpPacket* rtpPacket, uint32_t dataSize, char channel)
{

    rtpPacket->rtpHeader.seq = htons(rtpPacket->rtpHeader.seq);
    rtpPacket->rtpHeader.timestamp = htonl(rtpPacket->rtpHeader.timestamp);
    rtpPacket->rtpHeader.ssrc = htonl(rtpPacket->rtpHeader.ssrc);

    uint32_t rtpSize = RTP_HEADER_SIZE + dataSize;
    char* tempBuf = (char *)malloc(4 + rtpSize);
    tempBuf[0] = 0x24;//$
    tempBuf[1] = channel;// 0x00;
    tempBuf[2] = (uint8_t)(((rtpSize) & 0xFF00) >> 8);
    tempBuf[3] = (uint8_t)((rtpSize) & 0xFF);
    memcpy(tempBuf + 4, (char*)rtpPacket, rtpSize);

    int ret = send(clientSockfd, tempBuf, 4 + rtpSize, 0);

    rtpPacket->rtpHeader.seq = ntohs(rtpPacket->rtpHeader.seq);
    rtpPacket->rtpHeader.timestamp = ntohl(rtpPacket->rtpHeader.timestamp);
    rtpPacket->rtpHeader.ssrc = ntohl(rtpPacket->rtpHeader.ssrc);

    free(tempBuf);
    tempBuf = NULL;

    return ret;
}

音频传输

ret = rtpSendPacketOverTcp(clientSockfd, rtpPacket, frameSize + 4,0x02);

这里的音频流是0X02 视频流和音频流共有两路 RTP和RTCP占用了01 因为这是RTP传输的第二路流 所以是02

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static int rtpSendAACFrame(int clientSockfd,
    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 = rtpSendPacketOverTcp(clientSockfd, rtpPacket, frameSize + 4,0x02); 

    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;
}

2 3 字节 做运算 存储

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uint32_t rtpSize = RTP_HEADER_SIZE + dataSize;
char* tempBuf = (char *)malloc(4 + rtpSize);
tempBuf[0] = 0x24;//$ 分隔符
tempBuf[1] = channel;// 0x00; 通道
tempBuf[2] = (uint8_t)(((rtpSize) & 0xFF00) >> 8); //与运算后右移8位
tempBuf[3] = (uint8_t)((rtpSize) & 0xFF);
memcpy(tempBuf + 4, (char*)rtpPacket, rtpSize);

ffmpeg 命令行提取码流和音频文件

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//ffmpeg命令行 从mp4视频文件提取h264 码流文件
ffmpeg -i test.mp4 -an -vcodec copy -f h264 test.h264

//ffmpeg命令行 从mp4视频文件提取aac 音频文件
ffmpeg -i test.mp4  -vn -acodec aac test.aac