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调试libRTMP代码来分析RTMP协议

2017-07-25 22:13  jiayayao  阅读(3852)  评论(0编辑  收藏  举报

    RTMP是Real Time Messaging Protocol(实时消息传输协议)的首字母缩写。该协议基于TCP,是一个协议族,常用在视频直播领域。RTMP协议的默认端口是1935。

    学习一个协议最好的方法就是调试其通信过程,期间还可以使用wireshark抓包分析。本人在libRTMP的基础上添加了推流部分,并且使得整个过程变得可调试,学习其协议就变得简单多了。配置好的VS2010可调试的libRTMP工程:https://github.com/jiayayao/librtmp。该工程可以使用VS调试RTMP协议内部的代码,并且对RTMP协议部分做了详细的注释。推流部分参考leixiaohua的blog,RTMP Server可以采用Nginx-RTMP module的方式,搭建RTMP Server过程可以参考:使用nginx+nginx-rtmp-module+ffmpeg搭建流媒体服务器笔记(一)。 

    testRTMP工程是推流客户端,推送一个FLV文件需要经过以下几个步骤:握手,建立连接,建立流,推流。RTMP连接都是以握手作为开始的。建立连接阶段用于建立客户端与服务器之间的“网络连接”;建立流阶段用于建立客户端与服务器之间的“网络流”;推流即按照FLV格式将数据传送至RTMP Server。

一、握手

    RTMP握手过程如下:

1. 客户端向服务器发送C0、C1块,服务器收到后发送S0和S1块;

2. 客户端收到S0和S1后,向服务器发送C2块;服务器收到C2块后发送S2块;

3. 客户端和服务器分别收到S2和C2后,握手建立完成。

    与HandShake相对应,还有SHandShake函数是服务器部分的握手部分,有兴趣的可以看一下。

二、建立网络连接

三、建立网络流

    RTMP_ConnectStream()时接收到的packet的type依次是:

0x05: Set server bindwidth(BW = 50000000x06: Set client bindwidth(BW = 50000000x01: Set in chunk size(40960x20:Invoke <_result>
(object begin)
(object begin)
Property: <Name:             fmsVer, STRING:      FMS/3,0,1,123>
Property: <Name:       capabilities, NUMBER: 31.00>
(object end)
(object begin)
Property: <Name:              level, STRING: status>
Property: <Name:               code, STRING:      NetConnection.Connect.Success>
Property: <Name:        description, STRING:  Connection succeeded.>
Property: <Name:     objectEncoding, NUMBER:     0.00>
(object end)
(object end)

0x20:Invoke <_result>
(object begin)
Property: NULL
(object end) 

0x20:Invoke <onStatus>
(object begin)
Property: NULL
(object begin)
Property: <Name:              level, STRING: status>
Property: <Name:               code, STRING:      NetStream.Publish.Start>
Property: <Name:        description, STRING:  Start publishing>
(object end)
(object end)

    服务器接收到“connect”消息后,会返回_result给客户端,客户端接收到是connect的response后,会发送“createStream”命令到服务器。 

    服务器接收到“createStream”消息后,会返回_result给客户端,客户端接收到是“createStream”命令返回的response后,会发送“publish”命令到服务器。网络流建立完成,开始传送数据。

四、推流

    推流部分的关键代码如下:

int publish_using_packet(){
    RTMP *rtmp=NULL;            
    RTMPPacket *packet=NULL;
    uint32_t start_time=0;
    uint32_t now_time=0;
    //the timestamp of the previous frame
    long pre_frame_time=0; 
    long lasttime=0;
    int bNextIsKey=1;
    uint32_t preTagsize=0;

    //packet attributes
    uint32_t type=0;            
    uint32_t datalength=0;        
    uint32_t timestamp=0;        
    uint32_t streamid=0;            

    FILE*fp=NULL;
    fp=fopen("cuc_ieschool.flv","rb");
    if (!fp){
        RTMP_LogPrintf("Open File Error.\n");
        CleanupSockets();
        return -1;
    }

    if (!InitSockets()){
        RTMP_LogPrintf("Init Socket Err\n");
        return -1;
    }

    // 创建一个RTMP会话的句柄
    rtmp=RTMP_Alloc();
    // 初始化RTMP句柄
    RTMP_Init(rtmp);
    //set connection timeout,default 30s
    rtmp->Link.timeout=5;
    // 设置URL
    if(!RTMP_SetupURL(rtmp,"rtmp://192.168.37.130:1935/myapp/test1"))
    {
        RTMP_Log(RTMP_LOGERROR,"SetupURL Err\n");
        RTMP_Free(rtmp);
        CleanupSockets();
        return -1;
    }

    //if unable,the AMF command would be 'play' instead of 'publish'
    RTMP_EnableWrite(rtmp);    

    // RTMP_Connect分为2步:RTMP_Connect0和RTMP_Connect1
    // 0负责建立TCP底层连接
    // 1负责RTMP握手操作
    if (!RTMP_Connect(rtmp,NULL)){
        RTMP_Log(RTMP_LOGERROR,"Connect Err\n");
        RTMP_Free(rtmp);
        CleanupSockets();
        return -1;
    }

    if (!RTMP_ConnectStream(rtmp,0)){
        RTMP_Log(RTMP_LOGERROR,"ConnectStream Err\n");
        RTMP_Close(rtmp);
        RTMP_Free(rtmp);
        CleanupSockets();
        return -1;
    }

    packet=(RTMPPacket*)malloc(sizeof(RTMPPacket));
    RTMPPacket_Alloc(packet,1024*64);
    RTMPPacket_Reset(packet);

    packet->m_hasAbsTimestamp = 0;    
    packet->m_nChannel = 0x04;    
    packet->m_nInfoField2 = rtmp->m_stream_id;

    RTMP_LogPrintf("Start to send data ...\n");

    //jump over FLV Header
    // FLV格式的header为9个字节
    fseek(fp,9,SEEK_SET);    
    //jump over previousTagSizen
    // 跳过表征前一段Tag大小的4个字节
    fseek(fp,4,SEEK_CUR);    
    start_time=RTMP_GetTime();
    while(1)
    {
        if((((now_time=RTMP_GetTime())-start_time)
            <(pre_frame_time)) && bNextIsKey){    
                //wait for 1 sec if the send process is too fast
                //this mechanism is not very good,need some improvement
                if(pre_frame_time>lasttime){
                    RTMP_LogPrintf("TimeStamp:%8lu ms\n",pre_frame_time);
                    lasttime=pre_frame_time;
                }
                Sleep(1000);
                continue;
        }

        //not quite the same as FLV spec
        // 读取当前Tag的类型(1个字节)
        if(!ReadU8(&type,fp))    
            break;
        // 读取当前Tag data部分的大小(3个字节)
        if(!ReadU24(&datalength,fp))
            break;
        // 读取时间戳(4个字节)
        if(!ReadTime(&timestamp,fp))
            break;
        // 读取stream id(3个字节),一般为0
        if(!ReadU24(&streamid,fp))
            break;

        // 跳过既非视频也非音频的Tag
        if (type!=0x08&&type!=0x09){
            //jump over non_audio and non_video frame,
            //jump over next previousTagSizen at the same time
            fseek(fp,datalength+4,SEEK_CUR);
            continue;
        }

        // 读取当前音视频Tag的数据到packet
        if(fread(packet->m_body,1,datalength,fp)!=datalength)
            break;

        packet->m_headerType = RTMP_PACKET_SIZE_LARGE; 
        packet->m_nTimeStamp = timestamp; 
        packet->m_packetType = type;
        packet->m_nBodySize  = datalength;
        pre_frame_time=timestamp;

        if (!RTMP_IsConnected(rtmp)){
            RTMP_Log(RTMP_LOGERROR,"rtmp is not connect\n");
            break;
        }
        // 这样看下来是一个FLV的Tag发送一个RTMPPacket
        if (!RTMP_SendPacket(rtmp,packet,0)){
            RTMP_Log(RTMP_LOGERROR,"Send Error\n");
            break;
        }

        // 读取前一个Tag的size
        if(!ReadU32(&preTagsize,fp))
            break;

        // 读取当前Tag的type
        if(!PeekU8(&type,fp))
            break;
        if(type==0x09){
            if(fseek(fp,11,SEEK_CUR)!=0)
                break;
            if(!PeekU8(&type,fp)){
                break;
            }
            if(type==0x17)
                bNextIsKey=1;
            else
                bNextIsKey=0;

            fseek(fp,-11,SEEK_CUR);
        }
    }                

    RTMP_LogPrintf("\nSend Data Over\n");

    if(fp)
        fclose(fp);

    if (rtmp!=NULL){
        RTMP_Close(rtmp);    
        RTMP_Free(rtmp);    
        rtmp=NULL;
    }
    if (packet!=NULL){
        RTMPPacket_Free(packet);    
        free(packet);
        packet=NULL;
    }

    CleanupSockets();
    return 0;
}

     在推送过程中,打开VLC播放器,输入网络流地址为:"rtmp://192.168.37.130:1935/myapp/test1",即可看到推流客户端推送的视频。

参考资料:

1. RTMP流媒体播放过程

2. RTMP规范简单分析