本文档结合FFmpeg源代码，详细解释RTMP (Real-Time Messaging Protocol) 推流的完整流程和原理。

一、RTMP协议概述

1.1 RTMP简介

RTMP是Adobe公司开发的流媒体传输协议，主要用于Flash Player与服务器之间的音视频数据传输。

特点:

• 基于TCP协议，保证可靠传输
• 低延迟 (通常1-3秒)
• 支持实时互动
• 使用AMF (Action Message Format) 编码
• 采用分块 (Chunk) 传输机制

协议栈:

┌──────────────────────────────────┐
│     Application (FLV Data)       │
├──────────────────────────────────┤
│     RTMP Message (AMF/Data)      │
├──────────────────────────────────┤
│     RTMP Chunk (分块传输)         │
├──────────────────────────────────┤
│     TCP (可靠传输)                │
├──────────────────────────────────┤
│     IP                           │
└──────────────────────────────────┘

1.2 RTMP变种

位置: libavformat/rtmpproto.c:2695-2723

if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
    // RTMPT: RTMP over HTTP (隧道模式)
    ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
} else if (!strcmp(proto, "rtmps")) {
    // RTMPS: RTMP over TLS (加密)
    if (port < 0)
        port = RTMPS_DEFAULT_PORT;  // 443
    ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
} else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
    // RTMPE: RTMP Encrypted (加密)
    ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
    rt->encrypted = 1;
} else {
    // RTMP: 标准RTMP
    if (port < 0)
        port = RTMP_DEFAULT_PORT;  // 1935
    ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
}

协议对比:

协议	端口	传输层	加密	用途
rtmp	1935	TCP	否	标准RTMP
rtmps	443	TLS	是	加密RTMP
rtmpe	1935	TCP	是	Adobe加密
rtmpt	80	HTTP	否	穿透防火墙
rtmpte	80	HTTP	是	加密隧道

二、RTMP推流整体流程

2.1 流程图

┌─────────────────────────────────────────────────────────┐
│                    RTMP推流完整流程                       │
└─────────────────────────────────────────────────────────┘

1. TCP连接建立
   ├─→ 连接到服务器端口1935
   └─→ 建立TCP连接

2. RTMP握手 (Handshake)
   ├─→ C0+C1: 客户端发送版本和随机数据
   ├─→ S0+S1+S2: 服务器响应
   └─→ C2: 客户端确认

3. 连接 (Connect)
   ├─→ 发送connect命令
   ├─→ 携带app名称、flashVer等信息
   └─→ 接收_result响应

4. 创建流 (CreateStream)
   ├─→ 发送createStream命令
   └─→ 接收stream_id

5. 发布流 (Publish)
   ├─→ 可选: releaseStream
   ├─→ 可选: FCPublish
   ├─→ 发送publish命令
   └─→ 接收onStatus(NetStream.Publish.Start)

6. 数据传输
   ├─→ 发送音频数据包 (RTMP_PT_AUDIO)
   ├─→ 发送视频数据包 (RTMP_PT_VIDEO)
   ├─→ 发送元数据 (RTMP_PT_NOTIFY)
   └─→ 接收服务器控制消息

7. 结束流
   ├─→ 可选: FCUnpublish
   ├─→ 发送deleteStream
   └─→ 关闭TCP连接

2.2 状态机

位置: libavformat/rtmpproto.c:61-71

typedef enum {
    STATE_START,      ///< 客户端初始状态
    STATE_HANDSHAKED, ///< 完成握手
    STATE_FCPUBLISH,  ///< FCPublishing流 (推流)
    STATE_PLAYING,    ///< 开始接收数据 (拉流)
    STATE_SEEKING,    ///< 执行seek操作
    STATE_PUBLISHING, ///< 开始发送数据 (推流)
    STATE_RECEIVING,  ///< 收到publish命令 (服务端)
    STATE_SENDING,    ///< 收到play命令 (服务端)
    STATE_STOPPED,    ///< 广播已停止
} ClientState;

状态转换:

START
  │
  ▼
HANDSHAKED (握手完成)
  │
  ├─→ FCPUBLISH (可选)
  │     │
  │     ▼
  └─→ PUBLISHING (推流中)
        │
        ▼
      STOPPED

三、核心流程详解

3.1 TCP连接建立

位置: libavformat/rtmpproto.c:2726-2731

if ((ret = ffurl_open_whitelist(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
                                &s->interrupt_callback, opts,
                                s->protocol_whitelist, s->protocol_blacklist, s)) < 0) {
    av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
    goto fail;
}

关键点:

• 默认端口: 1935
• 支持TCP_NODELAY选项 (禁用Nagle算法)
• 支持连接超时设置

3.2 RTMP握手 (Handshake)

RTMP握手是一个三次握手过程，用于建立连接和验证客户端。

3.2.1 握手流程

位置: libavformat/rtmpproto.c:1243-1437

客户端                                服务器
   │                                     │
   │  C0 (1 byte: version=3)            │
   ├────────────────────────────────────▶│
   │                                     │
   │  C1 (1536 bytes: timestamp+random) │
   ├────────────────────────────────────▶│
   │                                     │
   │  S0 (1 byte: version=3)            │
   │◀────────────────────────────────────┤
   │                                     │
   │  S1 (1536 bytes: timestamp+random) │
   │◀────────────────────────────────────┤
   │                                     │
   │  S2 (1536 bytes: echo of C1)       │
   │◀────────────────────────────────────┤
   │                                     │
   │  C2 (1536 bytes: echo of S1)       │
   ├────────────────────────────────────▶│
   │                                     │
   │         握手完成                     │

3.2.2 握手包结构

C0/S0 (1字节):

 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|   Version=3   |
+-+-+-+-+-+-+-+-+

C1/S1 (1536字节):

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                        Time (4 bytes)                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                        Zero (4 bytes)                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                    Random bytes (1528 bytes)                  |
|                             ...                               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

3.2.3 握手代码实现

static int rtmp_handshake(URLContext *s, RTMPContext *rt)
{
    AVLFG rnd;
    uint8_t tosend[RTMP_HANDSHAKE_PACKET_SIZE+1] = {
        3,                // C0: 版本号
        0, 0, 0, 0,       // C1: 客户端时间
        RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
        RTMP_CLIENT_VER3, RTMP_CLIENT_VER4,
    };
    uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
    uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
    int ret;

    av_lfg_init(&rnd, 0xDEADC0DE);
    
    // 1. 生成C1随机数据 (1536字节)
    for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
        tosend[i] = av_lfg_get(&rnd) >> 24;

    // 2. 计算并添加digest (用于复杂握手)
    client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);

    // 3. 发送C0+C1
    if ((ret = ffurl_write(rt->stream, tosend,
                           RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
        av_log(s, AV_LOG_ERROR, "Cannot write RTMP handshake request\n");
        return ret;
    }

    // 4. 接收S0+S1
    if ((ret = ffurl_read_complete(rt->stream, serverdata,
                                   RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
        av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
        return ret;
    }

    // 5. 接收S2
    if ((ret = ffurl_read_complete(rt->stream, clientdata,
                                   RTMP_HANDSHAKE_PACKET_SIZE)) < 0) {
        av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
        return ret;
    }

    av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
           serverdata[5], serverdata[6], serverdata[7], serverdata[8]);

    // 6. 验证S1的digest
    if (rt->is_input && serverdata[5] >= 3) {
        server_pos = rtmp_validate_digest(serverdata + 1, 772);
        if (!server_pos) {
            server_pos = rtmp_validate_digest(serverdata + 1, 8);
            if (!server_pos) {
                av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
                return AVERROR(EIO);
            }
        }
        
        // 7. 计算C2的签名
        ret = ff_rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
                                  rtmp_server_key, sizeof(rtmp_server_key),
                                  digest);
        ret = ff_rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32,
                                  0, digest, 32, signature);
        
        // 8. 验证S2的签名
        if (memcmp(signature, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
            av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
            return AVERROR(EIO);
        }

        // 9. 生成并发送C2
        for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
            tosend[i] = av_lfg_get(&rnd) >> 24;
        
        ret = ff_rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
                                  rtmp_player_key, sizeof(rtmp_player_key),
                                  digest);
        ret = ff_rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
                                  digest, 32,
                                  tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
    }

    // 10. 发送C2
    if ((ret = ffurl_write(rt->stream, tosend,
                           RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
        return ret;

    return 0;
}

握手类型:

1. 简单握手: 不验证digest，直接echo
2. 复杂握手: 使用HMAC-SHA256验证，防止中间人攻击

3.3 连接 (Connect)

建立RTMP应用层连接，协商参数。

位置: libavformat/rtmpproto.c:323-450

static int gen_connect(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    // 1. 创建RTMP包
    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 4096 + APP_MAX_LENGTH)) < 0)
        return ret;

    p = pkt.data;

    // 2. 写入AMF数据
    ff_amf_write_string(&p, "connect");           // 命令名
    ff_amf_write_number(&p, ++rt->nb_invokes);    // 事务ID
    
    // 3. 写入连接参数对象
    ff_amf_write_object_start(&p);
    
    ff_amf_write_field_name(&p, "app");
    ff_amf_write_string2(&p, rt->app, rt->auth_params);  // 应用名称
    
    if (!rt->is_input) {
        ff_amf_write_field_name(&p, "type");
        ff_amf_write_string(&p, "nonprivate");    // 推流类型
    }
    
    ff_amf_write_field_name(&p, "flashVer");
    ff_amf_write_string(&p, rt->flashver);        // Flash版本
    
    ff_amf_write_field_name(&p, "tcUrl");
    ff_amf_write_string2(&p, rt->tcurl, rt->auth_params);  // 目标URL
    
    if (rt->is_input) {
        // 拉流时的能力声明
        ff_amf_write_field_name(&p, "fpad");
        ff_amf_write_bool(&p, 0);
        
        ff_amf_write_field_name(&p, "capabilities");
        ff_amf_write_number(&p, 15.0);
        
        ff_amf_write_field_name(&p, "audioCodecs");
        ff_amf_write_number(&p, 4071.0);  // 支持的音频编解码器
        
        ff_amf_write_field_name(&p, "videoCodecs");
        ff_amf_write_number(&p, 252.0);   // 支持的视频编解码器
        
        ff_amf_write_field_name(&p, "videoFunction");
        ff_amf_write_number(&p, 1.0);
    }
    
    ff_amf_write_object_end(&p);

    pkt.size = p - pkt.data;

    // 4. 发送包
    return rtmp_send_packet(rt, &pkt, 1);
}

Connect命令AMF格式:

┌──────────────────────────────────────┐
│         Connect命令结构               │
└──────────────────────────────────────┘

String: "connect"
Number: transaction_id (1.0)
Object: {
    app: "live"              // 应用名称
    type: "nonprivate"       // 推流类型
    flashVer: "FMLE/3.0"     // Flash版本
    tcUrl: "rtmp://server/live"  // 完整URL
    // 拉流时额外参数:
    fpad: false
    capabilities: 15.0
    audioCodecs: 4071.0
    videoCodecs: 252.0
    videoFunction: 1.0
}

服务器响应:

String: "_result"
Number: transaction_id (1.0)
Object: {
    fmsVer: "FMS/3,0,1,123"
    capabilities: 31.0
}
Object: {
    level: "status"
    code: "NetConnection.Connect.Success"
    description: "Connection succeeded."
    objectEncoding: 0.0
}

3.4 创建流 (CreateStream)

请求服务器分配一个流ID。

位置: libavformat/rtmpproto.c:715-733

static int gen_create_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    av_log(s, AV_LOG_DEBUG, "Creating stream...\n");

    // 1. 创建RTMP包
    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 25)) < 0)
        return ret;

    p = pkt.data;
    
    // 2. 写入AMF数据
    ff_amf_write_string(&p, "createStream");      // 命令名
    ff_amf_write_number(&p, ++rt->nb_invokes);    // 事务ID
    ff_amf_write_null(&p);                        // 命令对象 (null)

    // 3. 发送包
    return rtmp_send_packet(rt, &pkt, 1);
}

CreateStream命令格式:

String: "createStream"
Number: transaction_id (2.0)
Null

服务器响应:

String: "_result"
Number: transaction_id (2.0)
Null
Number: stream_id (1.0)  // 分配的流ID

3.5 发布流 (Publish)

通知服务器开始推流。

3.5.1 可选步骤

releaseStream (释放流):

static int gen_release_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 29 + strlen(rt->playpath))) < 0)
        return ret;

    av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
    p = pkt.data;
    ff_amf_write_string(&p, "releaseStream");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);  // 流名称

    return rtmp_send_packet(rt, &pkt, 1);
}

FCPublish (Flash Communication Publish):

static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 25 + strlen(rt->playpath))) < 0)
        return ret;

    av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
    p = pkt.data;
    ff_amf_write_string(&p, "FCPublish");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);

    return rtmp_send_packet(rt, &pkt, 1);
}

3.5.2 Publish命令

位置: libavformat/rtmpproto.c:887-909

static int gen_publish(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);

    // 1. 创建RTMP包
    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
                                     0, 30 + strlen(rt->playpath))) < 0)
        return ret;

    pkt.extra = rt->stream_id;  // 设置流ID

    p = pkt.data;
    
    // 2. 写入AMF数据
    ff_amf_write_string(&p, "publish");           // 命令名
    ff_amf_write_number(&p, ++rt->nb_invokes);    // 事务ID
    ff_amf_write_null(&p);                        // 命令对象
    ff_amf_write_string(&p, rt->playpath);        // 流名称
    ff_amf_write_string(&p, "live");              // 发布类型

    // 3. 发送包
    return rtmp_send_packet(rt, &pkt, 1);
}

Publish命令格式:

String: "publish"
Number: transaction_id (3.0)
Null
String: stream_name ("livestream")
String: publishing_type ("live" | "record" | "append")

发布类型:

• live: 实时流，不录制
• record: 录制流，覆盖同名文件
• append: 录制流，追加到文件

服务器响应:

String: "onStatus"
Number: 0.0
Null
Object: {
    level: "status"
    code: "NetStream.Publish.Start"
    description: "Stream is now published."
}

3.6 数据传输

3.6.1 数据包写入流程

位置: libavformat/rtmpproto.c:3014-3149

static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
    RTMPContext *rt = s->priv_data;
    int size_temp = size;
    int pktsize, pkttype, copy;
    uint32_t ts;
    const uint8_t *buf_temp = buf;
    int ret;

    do {
        // 1. 跳过FLV前一个包的尾部 (4字节)
        if (rt->skip_bytes) {
            int skip = FFMIN(rt->skip_bytes, size_temp);
            buf_temp       += skip;
            size_temp      -= skip;
            rt->skip_bytes -= skip;
            continue;
        }

        // 2. 读取FLV Tag Header (11字节)
        if (rt->flv_header_bytes < RTMP_HEADER) {
            copy = FFMIN(RTMP_HEADER - rt->flv_header_bytes, size_temp);
            bytestream_get_buffer(&buf_temp, 
                                 rt->flv_header + rt->flv_header_bytes, copy);
            rt->flv_header_bytes += copy;
            size_temp            -= copy;
            if (rt->flv_header_bytes < RTMP_HEADER)
                break;

            // 3. 解析FLV Tag Header
            const uint8_t *header = rt->flv_header;
            int channel = RTMP_AUDIO_CHANNEL;

            pkttype = bytestream_get_byte(&header);     // Tag类型
            pktsize = bytestream_get_be24(&header);     // 数据大小
            ts = bytestream_get_be24(&header);          // 时间戳低24位
            ts |= bytestream_get_byte(&header) << 24;   // 时间戳高8位
            bytestream_get_be24(&header);               // StreamID (总是0)
            
            rt->flv_size = pktsize;

            // 4. 确定RTMP通道
            if (pkttype == RTMP_PT_VIDEO)
                channel = RTMP_VIDEO_CHANNEL;

            // 5. 对于第一个音视频包或元数据，强制发送完整头
            if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
                pkttype == RTMP_PT_NOTIFY) {
                if ((ret = ff_rtmp_check_alloc_array(&rt->prev_pkt[1],
                                                     &rt->nb_prev_pkt[1],
                                                     channel)) < 0)
                    return ret;
                rt->prev_pkt[1][channel].channel_id = 0;
            }

            // 6. 创建RTMP包
            if ((ret = ff_rtmp_packet_create(&rt->out_pkt, channel,
                                             pkttype, ts, pktsize)) < 0)
                return ret;

            rt->out_pkt.extra = (rt->listen) ? rt->nb_streamid : rt->stream_id;
            rt->flv_data = rt->out_pkt.data;
        }

        // 7. 复制数据到RTMP包
        copy = FFMIN(rt->flv_size - rt->flv_off, size_temp);
        bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, copy);
        rt->flv_off += copy;
        size_temp   -= copy;

        // 8. 包完整时发送
        if (rt->flv_off == rt->flv_size) {
            rt->skip_bytes = 4;  // 下次跳过4字节 (PreviousTagSize)

            // 9. 处理元数据包 (添加@setDataFrame前缀)
            if (rt->out_pkt.type == RTMP_PT_NOTIFY) {
                uint8_t commandbuffer[64];
                int stringlen = 0;
                GetByteContext gbc;

                bytestream2_init(&gbc, rt->flv_data, rt->flv_size);
                if (!ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
                                        &stringlen)) {
                    if (!strcmp(commandbuffer, "onMetaData") ||
                        !strcmp(commandbuffer, "|RtmpSampleAccess")) {
                        uint8_t *ptr;
                        if ((ret = av_reallocp(&rt->out_pkt.data, 
                                              rt->out_pkt.size + 16)) < 0) {
                            rt->flv_size = rt->flv_off = rt->flv_header_bytes = 0;
                            return ret;
                        }
                        memmove(rt->out_pkt.data + 16, rt->out_pkt.data, 
                               rt->out_pkt.size);
                        rt->out_pkt.size += 16;
                        ptr = rt->out_pkt.data;
                        ff_amf_write_string(&ptr, "@setDataFrame");
                    }
                }
            }

            // 10. 发送RTMP包
            if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0)
                return ret;
            
            rt->flv_size = 0;
            rt->flv_off = 0;
            rt->flv_header_bytes = 0;
            rt->flv_nb_packets++;
        }
    } while (buf_temp - buf < size);

    // 11. 定期检查服务器消息
    if (rt->flv_nb_packets < rt->flush_interval)
        return size;
    rt->flv_nb_packets = 0;

    // 设置非阻塞模式
    rt->stream->flags |= AVIO_FLAG_NONBLOCK;
    ret = ffurl_read(rt->stream, &c, 1);
    rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;

    if (ret == 1) {
        RTMPPacket rpkt = { 0 };
        if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
                                                rt->in_chunk_size,
                                                &rt->prev_pkt[0],
                                                &rt->nb_prev_pkt[0], c)) <= 0)
             return ret;

        // 处理服务器消息
        if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
            return ret;

        ff_rtmp_packet_destroy(&rpkt);
    }

    return size;
}

3.6.2 FLV Tag格式

FLV Tag Header (11字节):

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|   TagType     |            DataSize (24 bits)                 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|            Timestamp (24 bits)                | TimestampExt  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|            StreamID (24 bits, always 0)                       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                         Tag Data                              |
|                            ...                                |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

TagType:

• 8: 音频
• 9: 视频
• 18: 脚本数据 (元数据)

音频Tag Data:

 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|F|  R  |S|  T  |  音频头
+-+-+-+-+-+-+-+-+
|   Audio Data  |  音频数据
|      ...      |
+-+-+-+-+-+-+-+-+

F: SoundFormat (4 bits)
   - 10: AAC
R: SoundRate (2 bits)
   - 0: 5.5kHz, 1: 11kHz, 2: 22kHz, 3: 44kHz
S: SoundSize (1 bit)
   - 0: 8bit, 1: 16bit
T: SoundType (1 bit)
   - 0: Mono, 1: Stereo

视频Tag Data:

 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|F|    C    | P |  视频头
+-+-+-+-+-+-+-+-+
|  Video Data   |  视频数据
|      ...      |
+-+-+-+-+-+-+-+-+

F: FrameType (4 bits)
   - 1: keyframe (I帧)
   - 2: inter frame (P帧)
   - 3: disposable inter frame (B帧)
C: CodecID (4 bits)
   - 7: AVC (H.264)
P: AVCPacketType (8 bits, 仅AVC)
   - 0: AVC sequence header (SPS/PPS)
   - 1: AVC NALU
   - 2: AVC end of sequence

四、RTMP Chunk协议

4.1 Chunk概念

RTMP将消息分割成更小的块 (Chunk) 进行传输，以支持多路复用。

特点:

• 默认chunk size: 128字节
• 可通过RTMP_PT_CHUNK_SIZE消息修改
• 不同通道的chunk可以交错发送
• 支持头部压缩

4.2 Chunk格式

位置: libavformat/rtmppkt.h:67-72

enum RTMPPacketSize {
    RTMP_PS_TWELVEBYTES = 0, ///< 12字节头 (完整头)
    RTMP_PS_EIGHTBYTES,      ///< 8字节头 (无Message Stream ID)
    RTMP_PS_FOURBYTES,       ///< 4字节头 (仅时间戳delta)
    RTMP_PS_ONEBYTE          ///< 1字节头 (续传chunk)
};

Chunk Header格式:

Type 0 (12字节) - 完整头:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|fmt|   cs id   |           timestamp (3 bytes)                 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           message length (3 bytes)            |message type id|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                   message stream id (4 bytes)                 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                   extended timestamp (4 bytes, if timestamp==0xFFFFFF)
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Type 1 (8字节) - 无Stream ID:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|fmt|   cs id   |       timestamp delta (3 bytes)               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           message length (3 bytes)            |message type id|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Type 2 (4字节) - 仅时间戳delta:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|fmt|   cs id   |       timestamp delta (3 bytes)               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Type 3 (1字节) - 续传chunk:

 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|fmt|   cs id   |
+-+-+-+-+-+-+-+-+

4.3 Chunk写入实现

位置: libavformat/rtmppkt.c:310-400

int ff_rtmp_packet_write(URLContext *h, RTMPPacket *pkt,
                         int chunk_size, RTMPPacket **prev_pkt_ptr,
                         int *nb_prev_pkt)
{
    uint8_t pkt_hdr[16], *p = pkt_hdr;
    int mode = RTMP_PS_TWELVEBYTES;
    int off = 0;
    int written = 0;
    int ret;
    RTMPPacket *prev_pkt;
    int use_delta;
    uint32_t timestamp;

    if ((ret = ff_rtmp_check_alloc_array(prev_pkt_ptr, nb_prev_pkt,
                                         pkt->channel_id)) < 0)
        return ret;
    prev_pkt = *prev_pkt_ptr;

    // 1. 判断是否可以使用delta时间戳
    use_delta = prev_pkt[pkt->channel_id].channel_id &&
        pkt->extra == prev_pkt[pkt->channel_id].extra &&
        pkt->timestamp >= prev_pkt[pkt->channel_id].timestamp;

    timestamp = pkt->timestamp;
    if (use_delta) {
        timestamp -= prev_pkt[pkt->channel_id].timestamp;
    }
    
    // 2. 处理扩展时间戳
    if (timestamp >= 0xFFFFFF) {
        pkt->ts_field = 0xFFFFFF;
    } else {
        pkt->ts_field = timestamp;
    }

    // 3. 确定chunk头类型
    if (use_delta) {
        if (pkt->type == prev_pkt[pkt->channel_id].type &&
            pkt->size == prev_pkt[pkt->channel_id].size) {
            mode = RTMP_PS_FOURBYTES;  // Type 2
            if (pkt->ts_field == prev_pkt[pkt->channel_id].ts_field)
                mode = RTMP_PS_ONEBYTE;  // Type 3
        } else {
            mode = RTMP_PS_EIGHTBYTES;  // Type 1
        }
    }

    // 4. 写入Basic Header
    if (pkt->channel_id < 64) {
        bytestream_put_byte(&p, pkt->channel_id | (mode << 6));
    } else if (pkt->channel_id < 64 + 256) {
        bytestream_put_byte(&p, 0 | (mode << 6));
        bytestream_put_byte(&p, pkt->channel_id - 64);
    } else {
        bytestream_put_byte(&p, 1 | (mode << 6));
        bytestream_put_le16(&p, pkt->channel_id - 64);
    }
    
    // 5. 写入Message Header
    if (mode != RTMP_PS_ONEBYTE) {
        bytestream_put_be24(&p, pkt->ts_field);
        if (mode != RTMP_PS_FOURBYTES) {
            bytestream_put_be24(&p, pkt->size);
            bytestream_put_byte(&p, pkt->type);
            if (mode == RTMP_PS_TWELVEBYTES) {
                bytestream_put_le32(&p, pkt->extra);
            }
        }
    }
    
    // 6. 写入扩展时间戳
    if (pkt->ts_field == 0xFFFFFF) {
        bytestream_put_be32(&p, timestamp);
    }

    // 7. 发送chunk头
    if (ffurl_write(h, pkt_hdr, p - pkt_hdr) < 0)
        return AVERROR(EIO);
    written = p - pkt_hdr + pkt->size;
    
    // 8. 分chunk发送数据
    while (off < pkt->size) {
        int towrite = FFMIN(chunk_size, pkt->size - off);
        if (ffurl_write(h, pkt->data + off, towrite) < 0)
            return AVERROR(EIO);
        off += towrite;
        
        // 9. 后续chunk使用Type 3头
        if (off < pkt->size) {
            uint8_t marker = 0xC0 | pkt->channel_id;
            if (ffurl_write(h, &marker, 1) < 0)
                return AVERROR(EIO);
            if (pkt->ts_field == 0xFFFFFF) {
                uint8_t ts_ext[4];
                bytestream_put_be32(&ts_ext, timestamp);
                if (ffurl_write(h, ts_ext, 4) < 0)
                    return AVERROR(EIO);
            }
        }
    }

    // 10. 保存历史信息
    prev_pkt[pkt->channel_id].channel_id = pkt->channel_id;
    prev_pkt[pkt->channel_id].type       = pkt->type;
    prev_pkt[pkt->channel_id].size       = pkt->size;
    prev_pkt[pkt->channel_id].timestamp  = pkt->timestamp;
    prev_pkt[pkt->channel_id].ts_field   = pkt->ts_field;
    prev_pkt[pkt->channel_id].extra      = pkt->extra;

    return written;
}

4.4 Chunk示例

发送一个1000字节的视频包 (chunk_size=128):

Chunk 1 (Type 0, 12字节头 + 128字节数据):
[0x06] [0x00 0x00 0x64] [0x00 0x03 0xE8] [0x09] [0x00 0x00 0x00 0x01]
[128 bytes data]

Chunk 2 (Type 3, 1字节头 + 128字节数据):
[0xC6]
[128 bytes data]

Chunk 3 (Type 3, 1字节头 + 128字节数据):
[0xC6]
[128 bytes data]

...

Chunk 8 (Type 3, 1字节头 + 104字节数据):
[0xC6]
[104 bytes data]

五、RTMP通道和消息类型

5.1 通道定义

位置: libavformat/rtmppkt.h:36-42

enum RTMPChannel {
    RTMP_NETWORK_CHANNEL = 2,   ///< 网络控制消息 (带宽报告、ping等)
    RTMP_SYSTEM_CHANNEL,        ///< 服务器控制消息 (connect、createStream等)
    RTMP_AUDIO_CHANNEL,         ///< 音频数据
    RTMP_VIDEO_CHANNEL   = 6,   ///< 视频数据
    RTMP_SOURCE_CHANNEL  = 8,   ///< 音视频调用 (publish、play等)
};

5.2 消息类型

位置: libavformat/rtmppkt.h:47-62

typedef enum RTMPPacketType {
    RTMP_PT_CHUNK_SIZE   =  1,  ///< 设置chunk大小
    RTMP_PT_BYTES_READ   =  3,  ///< 已读字节数确认
    RTMP_PT_USER_CONTROL,       ///< 用户控制消息
    RTMP_PT_WINDOW_ACK_SIZE,    ///< 窗口确认大小
    RTMP_PT_SET_PEER_BW,        ///< 设置对等带宽
    RTMP_PT_AUDIO        =  8,  ///< 音频包
    RTMP_PT_VIDEO,              ///< 视频包
    RTMP_PT_FLEX_STREAM  = 15,  ///< Flex共享流
    RTMP_PT_FLEX_OBJECT,        ///< Flex共享对象
    RTMP_PT_FLEX_MESSAGE,       ///< Flex共享消息
    RTMP_PT_NOTIFY,             ///< 通知 (元数据)
    RTMP_PT_SHARED_OBJ,         ///< 共享对象
    RTMP_PT_INVOKE,             ///< 调用 (命令)
    RTMP_PT_METADATA     = 22,  ///< FLV元数据
} RTMPPacketType;

5.3 控制消息

Chunk Size (Type 1):

static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;
    if (pkt->size < 4) {
        av_log(s, AV_LOG_ERROR, "Too short chunk size change packet (%d)\n",
               pkt->size);
        return AVERROR_INVALIDDATA;
    }

    rt->in_chunk_size = AV_RB32(pkt->data);
    if (rt->in_chunk_size <= 0) {
        av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n",
               rt->in_chunk_size);
        return AVERROR_INVALIDDATA;
    }
    av_log(s, AV_LOG_DEBUG, "New incoming chunk size = %d\n",
           rt->in_chunk_size);
    return 0;
}

Window Acknowledgement Size (Type 5):

static int handle_window_ack_size(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;
    if (pkt->size < 4) {
        av_log(s, AV_LOG_ERROR,
               "Too short window acknowledgement size packet (%d)\n",
               pkt->size);
        return AVERROR_INVALIDDATA;
    }

    rt->receive_report_size = AV_RB32(pkt->data);
    if (rt->receive_report_size <= 0) {
        av_log(s, AV_LOG_ERROR, "Incorrect window acknowledgement size %d\n",
               rt->receive_report_size);
        return AVERROR_INVALIDDATA;
    }
    av_log(s, AV_LOG_DEBUG, "Window acknowledgement size = %d\n",
           rt->receive_report_size);
    return 0;
}

Set Peer Bandwidth (Type 6):

static int handle_set_peer_bw(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;
    if (pkt->size < 4) {
        av_log(s, AV_LOG_ERROR,
               "Too short set peer bandwidth packet (%d)\n",
               pkt->size);
        return AVERROR_INVALIDDATA;
    }

    rt->max_sent_unacked = AV_RB32(pkt->data);
    av_log(s, AV_LOG_DEBUG, "Max sent, unacked = %d\n",
           rt->max_sent_unacked);
    return 0;
}

六、AMF编码

6.1 AMF简介

AMF (Action Message Format) 是Adobe的二进制数据格式，用于序列化对象。

AMF0数据类型:

#define AMF_DATA_TYPE_NUMBER      0x00
#define AMF_DATA_TYPE_BOOL        0x01
#define AMF_DATA_TYPE_STRING      0x02
#define AMF_DATA_TYPE_OBJECT      0x03
#define AMF_DATA_TYPE_NULL        0x05
#define AMF_DATA_TYPE_UNDEFINED   0x06
#define AMF_DATA_TYPE_ARRAY       0x08
#define AMF_DATA_TYPE_OBJECT_END  0x09
#define AMF_DATA_TYPE_MIXEDARRAY  0x0A

6.2 AMF编码示例

Number (8字节):

[0x00] [IEEE 754 double, 8 bytes]

String:

[0x02] [length, 2 bytes] [string data]

Object:

[0x03]
  [property name length, 2 bytes] [property name]
  [property value type] [property value]
  ...
[0x00 0x00 0x09]  // Object End

Null:

[0x05]

6.3 Connect命令AMF示例

02 00 07 "connect"           // String: "connect"
00 3F F0 00 00 00 00 00 00   // Number: 1.0
03                            // Object start
  00 03 "app"                 // Property: "app"
  02 00 04 "live"             // String: "live"
  00 04 "type"                // Property: "type"
  02 00 0A "nonprivate"       // String: "nonprivate"
  00 08 "flashVer"            // Property: "flashVer"
  02 00 0E "FMLE/3.0 ..."     // String: "FMLE/3.0..."
  00 05 "tcUrl"               // Property: "tcUrl"
  02 00 1F "rtmp://..."       // String: "rtmp://..."
00 00 09                      // Object end

七、完整推流示例

7.1 推流命令

ffmpeg -re -i input.mp4 -c copy -f flv rtmp://server/live/stream

7.2 网络交互时序

客户端                                    服务器
  │                                         │
  │  1. TCP SYN                             │
  ├────────────────────────────────────────▶│
  │  2. TCP SYN-ACK                         │
  │◀────────────────────────────────────────┤
  │  3. TCP ACK                             │
  ├────────────────────────────────────────▶│
  │                                         │
  │  4. C0+C1 (Handshake)                   │
  ├────────────────────────────────────────▶│
  │  5. S0+S1+S2                            │
  │◀────────────────────────────────────────┤
  │  6. C2                                  │
  ├────────────────────────────────────────▶│
  │                                         │
  │  7. Set Chunk Size (4096)               │
  ├────────────────────────────────────────▶│
  │  8. Connect("live")                     │
  ├────────────────────────────────────────▶│
  │  9. Window Ack Size                     │
  │◀────────────────────────────────────────┤
  │  10. Set Peer Bandwidth                 │
  │◀────────────────────────────────────────┤
  │  11. User Control (Stream Begin)        │
  │◀────────────────────────────────────────┤
  │  12. Set Chunk Size                     │
  │◀────────────────────────────────────────┤
  │  13. _result(NetConnection.Connect.Success)│
  │◀────────────────────────────────────────┤
  │  14. onBWDone                           │
  │◀────────────────────────────────────────┤
  │                                         │
  │  15. releaseStream("stream")            │
  ├────────────────────────────────────────▶│
  │  16. FCPublish("stream")                │
  ├────────────────────────────────────────▶│
  │  17. createStream()                     │
  ├────────────────────────────────────────▶│
  │  18. _result(stream_id=1)               │
  │◀────────────────────────────────────────┤
  │                                         │
  │  19. publish("stream", "live")          │
  ├────────────────────────────────────────▶│
  │  20. onStatus(NetStream.Publish.Start)  │
  │◀────────────────────────────────────────┤
  │                                         │
  │  21. @setDataFrame("onMetaData", {...}) │
  ├────────────────────────────────────────▶│
  │  22. Video (AVC Sequence Header)        │
  ├────────────────────────────────────────▶│
  │  23. Audio (AAC Sequence Header)        │
  ├────────────────────────────────────────▶│
  │  24. Video (Keyframe)                   │
  ├────────────────────────────────────────▶│
  │  25. Audio Data                         │
  ├────────────────────────────────────────▶│
  │  26. Video (P-frame)                    │
  ├────────────────────────────────────────▶│
  │  ...                                    │
  │  (持续发送音视频数据)                    │
  │  ...                                    │
  │                                         │
  │  N. deleteStream(stream_id)             │
  ├────────────────────────────────────────▶│
  │  N+1. TCP FIN                           │
  ├────────────────────────────────────────▶│

7.3 元数据示例

{
    "duration": 0,
    "width": 1920,
    "height": 1080,
    "videodatarate": 2500,
    "framerate": 30,
    "videocodecid": 7,  // AVC
    "audiodatarate": 128,
    "audiosamplerate": 48000,
    "audiosamplesize": 16,
    "stereo": true,
    "audiocodecid": 10,  // AAC
    "encoder": "Lavf58.76.100"
}

八、关键数据结构

8.1 RTMPContext

位置: libavformat/rtmpproto.c:79-137

typedef struct RTMPContext {
    const AVClass *class;
    URLContext*   stream;                     ///< TCP流
    RTMPPacket    *prev_pkt[2];               ///< 包历史 ([0]读, [1]写)
    int           nb_prev_pkt[2];             ///< prev_pkt元素数
    int           in_chunk_size;              ///< 接收chunk大小
    int           out_chunk_size;             ///< 发送chunk大小
    int           is_input;                   ///< 输入/输出标志
    char          *playpath;                  ///< 流标识符
    int           live;                       ///< 0:录制, -1:直播, -2:两者
    char          *app;                       ///< 应用名称
    char          *conn;                      ///< 附加AMF数据
    ClientState   state;                      ///< 当前状态
    int           stream_id;                  ///< 服务器分配的流ID
    uint8_t*      flv_data;                   ///< demuxer缓冲区
    int           flv_size;                   ///< 当前缓冲区大小
    int           flv_off;                    ///< 已读字节数
    int           flv_nb_packets;             ///< 已发布的flv包数
    RTMPPacket    out_pkt;                    ///< 输出rtmp包
    uint32_t      receive_report_size;        ///< 报告接收字节数的阈值
    uint64_t      bytes_read;                 ///< 从服务器读取的字节数
    uint64_t      last_bytes_read;            ///< 上次报告的字节数
    uint32_t      last_timestamp;             ///< 最后接收包的时间戳
    int           skip_bytes;                 ///< 下次写入时跳过的字节数
    int           has_audio;                  ///< 是否有音频
    int           has_video;                  ///< 是否有视频
    int           received_metadata;          ///< 是否接收到元数据
    uint8_t       flv_header[RTMP_HEADER];    ///< 部分flv包头
    int           flv_header_bytes;           ///< flv_header中已初始化字节数
    int           nb_invokes;                 ///< invoke消息计数
    char*         tcurl;                      ///< 目标流URL
    char*         flashver;                   ///< flash插件版本
    char*         swfurl;                     ///< swf播放器URL
    char*         pageurl;                    ///< 网页URL
    int           max_sent_unacked;           ///< 最大未确认发送字节数
    int           client_buffer_time;         ///< 客户端缓冲时间(ms)
    int           flush_interval;             ///< 刷新间隔(包数)
    int           encrypted;                  ///< 使用加密连接(RTMPE)
    TrackedMethod*tracked_methods;            ///< 跟踪的方法缓冲区
    int           nb_tracked_methods;         ///< 跟踪的方法数
    int           listen;                     ///< 监听模式标志
    int           tcp_nodelay;                ///< 使用TCP_NODELAY
} RTMPContext;

8.2 RTMPPacket

位置: libavformat/rtmppkt.h:77-87

typedef struct RTMPPacket {
    int            channel_id;    ///< 通道ID
    RTMPPacketType type;          ///< 包类型
    uint32_t       timestamp;     ///< 完整时间戳
    uint32_t       ts_field;      ///< 24位时间戳或增量(ms)
    uint32_t       extra;         ///< 额外的通道ID
    uint8_t        *data;         ///< 包payload
    int            size;          ///< payload大小
    int            offset;        ///< 已读数据量
    int            read;          ///< 已读总量(包括头)
} RTMPPacket;

九、性能优化和最佳实践

9.1 Chunk Size优化

// 默认chunk size较小 (128字节)，增大可以减少头部开销
rt->out_chunk_size = 4096;  // 推荐值

// 发送Set Chunk Size消息
RTMPPacket pkt;
ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, 
                      RTMP_PT_CHUNK_SIZE, 0, 4);
uint8_t *p = pkt.data;
bytestream_put_be32(&p, rt->out_chunk_size);
rtmp_send_packet(rt, &pkt, 0);

9.2 TCP_NODELAY

// 禁用Nagle算法，减少延迟
rt->tcp_nodelay = 1;

9.3 缓冲区管理

// 客户端缓冲时间设置
rt->client_buffer_time = 3000;  // 3秒

// 发送Buffer Time消息
gen_buffer_time(s, rt);

9.4 错误处理

// 定期检查服务器消息
if (rt->flv_nb_packets >= rt->flush_interval) {
    rt->flv_nb_packets = 0;
    
    // 非阻塞读取
    rt->stream->flags |= AVIO_FLAG_NONBLOCK;
    ret = ffurl_read(rt->stream, &c, 1);
    rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
    
    if (ret == 1) {
        // 处理服务器消息
        rtmp_parse_result(s, rt, &rpkt);
    }
}

十、常见问题和调试

10.1 连接失败

问题: 无法连接到RTMP服务器

排查:

1. 检查网络连通性
2. 验证URL格式: rtmp://server[:port]/app/stream
3. 检查防火墙设置
4. 查看服务器日志

10.2 握手失败

问题: Handshake验证失败

原因:

• 服务器要求复杂握手，但客户端不支持
• 加密握手配置错误

解决:

# 使用简单握手
ffmpeg -rtmp_enhanced_codecs "" -i input.mp4 -f flv rtmp://server/live/stream

10.3 推流卡顿

问题: 推流过程中出现卡顿

原因:

• 网络带宽不足
• Chunk size太小
• 服务器处理能力不足

解决:

# 降低码率
ffmpeg -i input.mp4 -b:v 1000k -b:a 128k -f flv rtmp://server/live/stream

# 增大chunk size (在代码中设置)
rt->out_chunk_size = 8192;

10.4 调试技巧

启用详细日志:

ffmpeg -loglevel debug -i input.mp4 -f flv rtmp://server/live/stream

抓包分析:

tcpdump -i any -w rtmp.pcap port 1935
wireshark rtmp.pcap

代码调试:

// 在关键位置添加日志
av_log(s, AV_LOG_DEBUG, "Sending packet: type=%d, size=%d, ts=%u\n",
       pkt.type, pkt.size, pkt.timestamp);

// 启用包转储
#define DEBUG
ff_rtmp_packet_dump(s, &pkt);

十一、总结

RTMP推流的完整流程:

1. TCP连接: 建立到服务器的TCP连接
2. 握手: 三次握手建立RTMP连接
3. 连接: 发送connect命令，协商参数
4. 创建流: 发送createStream，获取stream_id
5. 发布: 发送publish命令，开始推流
6. 数据传输:
   • 发送元数据 (@setDataFrame)
   • 发送AVC/AAC Sequence Header
   • 持续发送音视频数据
7. 结束: 发送deleteStream，关闭连接

关键特性:

• 分块传输: Chunk机制支持多路复用
• 头部压缩: 4种chunk头类型减少开销
• AMF编码: 高效的二进制序列化
• 低延迟: 1-3秒延迟，适合实时互动
• 可靠传输: 基于TCP，保证数据完整性

适用场景:

• 直播推流
• 视频会议
• 在线教育
• 游戏直播