SRS流媒体服务器：RTMP推流、拉流创建连接

目录

1. 识别客户端，然后获取或者创建SrsLiveSource
2. 启动推流
3. 启动拉流

1. RTMP推流、拉流创建连接

1.1RTMP推流和拉流创建连接对象的⽅式都是创建了SrsRtmpConn，见上面SrsServer::fd_to_resource函数。

1.2每个SrsRtmpConn都绑定⼀个SrsCoroutine，具体的业务处理在SrsCoroutine的循环进⾏，对于RTMP⽽⾔最终的循环为SrsRtmpConn::cycle函数。

1.3而SrsRtmpConn::cycle函数最终会调用SrsRtmpConn::do_cycle函数。

srs_error_t SrsRtmpConn::do_cycle()
{
    srs_error_t err = srs_success;
  
    srs_trace("RTMP client ip=%s:%d, fd=%d", ip.c_str(), port, srs_netfd_fileno(stfd));
  
    rtmp->set_recv_timeout(SRS_CONSTS_RTMP_TIMEOUT);
    rtmp->set_send_timeout(SRS_CONSTS_RTMP_TIMEOUT);

    if ((err = rtmp->handshake()) != srs_success) { //RTMP握手逻辑
        return srs_error_wrap(err, "rtmp handshake");
    }

    uint32_t rip = rtmp->proxy_real_ip();
    if (rip > 0) {
        srs_trace("RTMP proxy real client ip=%d.%d.%d.%d",
            uint8_t(rip>>24), uint8_t(rip>>16), uint8_t(rip>>8), uint8_t(rip));
    }
  
    SrsRequest* req = info->req;
    if ((err = rtmp->connect_app(req)) != srs_success) { //接收connect请求
        return srs_error_wrap(err, "rtmp connect tcUrl");
    }
  
    // set client ip to request.
    req->ip = ip;
  
    srs_trace("connect app, tcUrl=%s, pageUrl=%s, swfUrl=%s, schema=%s, vhost=%s, port=%d, app=%s, args=%s",
        req->tcUrl.c_str(), req->pageUrl.c_str(), req->swfUrl.c_str(),
        req->schema.c_str(), req->vhost.c_str(), req->port,
        req->app.c_str(), (req->args? "(obj)":"null"));
  
    ...
  
    if ((err = service_cycle()) != srs_success) {
        err = srs_error_wrap(err, "service cycle");
    }
  
    srs_error_t r0 = srs_success;
    if ((r0 = on_disconnect()) != srs_success) {
        err = srs_error_wrap(err, "on disconnect %s", srs_error_desc(r0).c_str());
        srs_freep(r0);
    }
  
    // If client is redirect to other servers, we already logged the event.
    if (srs_error_code(err) == ERROR_CONTROL_REDIRECT) {
        srs_error_reset(err);
    }
  
    return err;
}

1.4SrsRtmpConn::cycle函数主要是进行RTMP握手，接收connect请求，判断为有效连接后调用SrsRtmpConn::service_cycle函数。

1.4.1todo：后续会写一篇具体RTMP握手过程，命令控制消息和协议控制消息交互逻辑文章。

srs_error_t SrsRtmpConn::service_cycle()
{
    srs_error_t err = srs_success;
  
    SrsRequest* req = info->req;
  
    int out_ack_size = _srs_config->get_out_ack_size(req->vhost);
    if (out_ack_size && (err = rtmp->set_window_ack_size(out_ack_size)) != srs_success) {
        return srs_error_wrap(err, "rtmp: set out window ack size");
    }
  
    int in_ack_size = _srs_config->get_in_ack_size(req->vhost);
    if (in_ack_size && (err = rtmp->set_in_window_ack_size(in_ack_size)) != srs_success) {
        return srs_error_wrap(err, "rtmp: set in window ack size");
    }
  
    if ((err = rtmp->set_peer_bandwidth((int)(2.5 * 1000 * 1000), 2)) != srs_success) {
        return srs_error_wrap(err, "rtmp: set peer bandwidth");
    }
  
    // get the ip which client connected.
    std::string local_ip = srs_get_local_ip(srs_netfd_fileno(stfd));
  
    // do bandwidth test if connect to the vhost which is for bandwidth check.
    if (_srs_config->get_bw_check_enabled(req->vhost)) {
        if ((err = bandwidth->bandwidth_check(rtmp, skt, req, local_ip)) != srs_success) {
            return srs_error_wrap(err, "rtmp: bandwidth check");
        }
        return err;
    }
  
    // set chunk size to larger.
    // set the chunk size before any larger response greater than 128,
    // to make OBS happy, @see https://github.com/ossrs/srs/issues/454
    int chunk_size = _srs_config->get_chunk_size(req->vhost);
    if ((err = rtmp->set_chunk_size(chunk_size)) != srs_success) {
        return srs_error_wrap(err, "rtmp: set chunk size %d", chunk_size);
    }
  
    // response the client connect ok. 响应客户端connect请求
    if ((err = rtmp->response_connect_app(req, local_ip.c_str())) != srs_success) {
        return srs_error_wrap(err, "rtmp: response connect app");
    }
  
    if ((err = rtmp->on_bw_done()) != srs_success) {
        return srs_error_wrap(err, "rtmp: on bw down");
    }
  
    while (true) {
        if ((err = trd->pull()) != srs_success) {
            return srs_error_wrap(err, "rtmp: thread quit");
        }
      
        err = stream_service_cycle();
      
        // stream service must terminated with error, never success.
        // when terminated with success, it's user required to stop.
        // TODO: FIXME: Support RTMP client timeout, https://github.com/ossrs/srs/issues/1134
        if (err == srs_success) {
            continue;
        }
      
        ...
      
        // for other system control message, fata