SRS流媒体服务器——RTMP推流、拉流创建连接

SRS安装部署相关内容：

SRS流媒体服务器——单机环境搭建和源码目录介绍

SRS流媒体服务器——Forward集群搭建和源码分析

SRS流媒体服务器——Edge集群搭建

SRS流媒体服务器——WebRTC推拉流演示

SRS流媒体服务器——SRS4.0 WebRTC⼀对⼀通话环境搭建与逻辑分析

SRS部分源码分析相关内容（请按照如下顺序阅读）：

SRS流媒体服务器——基本流程简单分析

SRS流媒体服务器——RTMP端⼝监听逻辑分析

SRS流媒体服务器——RTMP推流、拉流创建连接

SRS流媒体服务器——服务器读取RTMP推流数据

SRS流媒体服务器——服务器给RTMP拉流端转发数据

1. RTMP推流、拉流创建连接

RTMP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。
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。
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;
}

SrsRtmpConn::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. todo：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/454int 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/1134if (err == srs_success) {continue;}...// for other system control message, fatal error.return srs_error_wrap(err, "rtmp: reject");}return err;
}

设置协议控制消息后进入SrsRtmpConn::stream_service_cycle，stream_service_cycle负责流服务循环
1. 首先需要识别客户端，然后获取或者创建SrsLiveSource。
2. 然后根据SrsRtmpConnType判断推流还是拉流，执行推流逻辑（SrsRtmpConn::playing）或者拉流逻辑（SrsRtmpConn::publishing）

srs_error_t SrsRtmpConn::stream_service_cycle()
{srs_error_t err = srs_success;SrsRequest* req = info->req;//识别客户端if ((err = rtmp->identify_client(info->res->stream_id, info->type, req->stream, req->duration)) != srs_success) {return srs_error_wrap(err, "rtmp: identify client");}srs_discovery_tc_url(req->tcUrl, req->schema, req->host, req->vhost, req->app, req->stream, req->port, req->param);req->strip();srs_trace("client identified, type=%s, vhost=%s, app=%s, stream=%s, param=%s, duration=%dms",srs_client_type_string(info->type).c_str(), req->vhost.c_str(), req->app.c_str(), req->stream.c_str(), req->param.c_str(), srsu2msi(req->duration));// discovery vhost, resolve the vhost from configSrsConfDirective* parsed_vhost = _srs_config->get_vhost(req->vhost);if (parsed_vhost) {req->vhost = parsed_vhost->arg0();}if (req->schema.empty() || req->vhost.empty() || req->port == 0 || req->app.empty()) {return srs_error_new(ERROR_RTMP_REQ_TCURL, "discovery tcUrl failed, tcUrl=%s, schema=%s, vhost=%s, port=%d, app=%s",req->tcUrl.c_str(), req->schema.c_str(), req->vhost.c_str(), req->port, req->app.c_str());}// check vhost, allow default vhost.if ((err = check_vhost(true)) != srs_success) {return srs_error_wrap(err, "check vhost");}srs_trace("connected stream, tcUrl=%s, pageUrl=%s, swfUrl=%s, schema=%s, vhost=%s, port=%d, app=%s, stream=%s, param=%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->stream.c_str(), req->param.c_str(), (req->args? "(obj)":"null"));// do token traverse before serve it.// @see https://github.com/ossrs/srs/pull/239if (true) {info->edge = _srs_config->get_vhost_is_edge(req->vhost);bool edge_traverse = _srs_config->get_vhost_edge_token_traverse(req->vhost);if (info->edge && edge_traverse) {if ((err = check_edge_token_traverse_auth()) != srs_success) {return srs_error_wrap(err, "rtmp: check token traverse");}}}// security checkif ((err = security->check(info->type, ip, req)) != srs_success) {return srs_error_wrap(err, "rtmp: security check");}// Never allow the empty stream name, for HLS may write to a file with empty name.// @see https://github.com/ossrs/srs/issues/834if (req->stream.empty()) {return srs_error_new(ERROR_RTMP_STREAM_NAME_EMPTY, "rtmp: empty stream");}// client is identified, set the timeout to service timeout.rtmp->set_recv_timeout(SRS_CONSTS_RTMP_TIMEOUT);rtmp->set_send_timeout(SRS_CONSTS_RTMP_TIMEOUT);// find a source to serve.SrsLiveSource* source = NULL;if ((err = _srs_sources->fetch_or_create(req, server, &source)) != srs_success) { //获取或者创建SrsLiveSourcereturn srs_error_wrap(err, "rtmp: fetch source");}srs_assert(source != NULL);bool enabled_cache = _srs_config->get_gop_cache(req->vhost);srs_trace("source url=%s, ip=%s, cache=%d, is_edge=%d, source_id=%s/%s",req->get_stream_url().c_str(), ip.c_str(), enabled_cache, info->edge, source->source_id().c_str(), source->pre_source_id().c_str());source->set_cache(enabled_cache);switch (info->type) {case SrsRtmpConnPlay: {// response connection start playif ((err = rtmp->start_play(info->res->stream_id)) != srs_success) {return srs_error_wrap(err, "rtmp: start play");}if ((err = http_hooks_on_play()) != srs_success) {return srs_error_wrap(err, "rtmp: callback on play");}err = playing(source);http_hooks_on_stop();return err;}case SrsRtmpConnFMLEPublish: {if ((err = rtmp->start_fmle_publish(info->res->stream_id)) != srs_success) {return srs_error_wrap(err, "rtmp: start FMLE publish");}return publishing(source);}case SrsRtmpConnHaivisionPublish: {if ((err = rtmp->start_haivision_publish(info->res->stream_id)) != srs_success) {return srs_error_wrap(err, "rtmp: start HAIVISION publish");}return publishing(source);}case SrsRtmpConnFlashPublish: {if ((err = rtmp->start_flash_publish(info->res->stream_id)) != srs_success) {return srs_error_wrap(err, "rtmp: start FLASH publish");}return publishing(source);}default: {return srs_error_new(ERROR_SYSTEM_CLIENT_INVALID, "rtmp: unknown client type=%d", info->type);}}return err;
}

1. 识别客户端，然后获取或者创建SrsLiveSource

SrsRtmpServer::identify_client负责识别客户端。
1. 当协议控制消息是createStream时，返回SrsFMLEStartPacket对象，进行推流。
2. 当协议控制消息是play时，返回SrsPlayPacket对象，进行拉流。

srs_error_t SrsRtmpServer::identify_client(int stream_id, SrsRtmpConnType& type, string& stream_name, srs_utime_t& duration)
{type = SrsRtmpConnUnknown;srs_error_t err = srs_success;while (true) {SrsCommonMessage* msg = NULL;if ((err = protocol->recv_message(&msg)) != srs_success) {return srs_error_wrap(err, "recv identify message");}SrsAutoFree(SrsCommonMessage, msg);SrsMessageHeader& h = msg->header;if (h.is_ackledgement() || h.is_set_chunk_size() || h.is_window_ackledgement_size() || h.is_user_control_message()) {continue;}if (!h.is_amf0_command() && !h.is_amf3_command()) {srs_trace("ignore message type=%#x", h.message_type);continue;}SrsPacket* pkt = NULL;if ((err = protocol->decode_message(msg, &pkt)) != srs_success) {return srs_error_wrap(err, "decode identify");}SrsAutoFree(SrsPacket, pkt);if (dynamic_cast<SrsCreateStreamPacket*>(pkt)) {return identify_create_stream_client(dynamic_cast<SrsCreateStreamPacket*>(pkt), stream_id, 3, type, stream_name, duration);}if (dynamic_cast<SrsFMLEStartPacket*>(pkt)) { //RTMP推流会先发送releaseStream协议控制消息，解析后返回SrsFMLEStartPacket对象return identify_fmle_publish_client(dynamic_cast<SrsFMLEStartPacket*>(pkt), type, stream_name);}if (dynamic_cast<SrsPlayPacket*>(pkt)) { //RTMP拉流收到play协议控制消息，解析后返回SrsPlayPacket对象return identify_play_client(dynamic_cast<SrsPlayPacket*>(pkt), type, stream_name, duration);}// call msg,// support response null first,// @see https://github.com/ossrs/srs/issues/106// TODO: FIXME: response in right way, or forward in edge mode.SrsCallPacket* call = dynamic_cast<SrsCallPacket*>(pkt);if (call) {SrsCallResPacket* res = new SrsCallResPacket(call->transaction_id);res->command_object = SrsAmf0Any::null();res->response = SrsAmf0Any::null();if ((err = protocol->send_and_free_packet(res, 0)) != srs_success) {return srs_error_wrap(err, "response call");}// For encoder of Haivision, it always send a _checkbw call message.// @remark the next message is createStream, so we continue to identify it.// @see https://github.com/ossrs/srs/issues/844if (call->command_name == "_checkbw") {continue;}continue;}srs_trace("ignore AMF0/AMF3 command message.");}return err;
}

SrsProtocol::do_decode_message负责解析各种消息格式，例如上述的SrsFMLEStartPacket对象，解析代码是：

srs_error_t SrsProtocol::do_decode_message(SrsMessageHeader& header, SrsBuffer* stream, SrsPacket** ppacket)
{srs_error_t err = srs_success;SrsPacket* packet = NULL;// decode specified packet typeif (header.is_amf0_command() || header.is_amf3_command() || header.is_amf0_data() || header.is_amf3_data()) {// skip 1bytes to decode the amf3 command.if (header.is_amf3_command() && stream->require(1)) {stream->skip(1);}...// reset to zero(amf3 to 1) to restart decode.stream->skip(-1 * stream->pos());if (header.is_amf3_command()) {stream->skip(1);}// decode command object.if (command == RTMP_AMF0_COMMAND_CONNECT) {*ppacket = packet = new SrsConnectAppPacket();return packet->decode(stream);} else if (command == RTMP_AMF0_COMMAND_CREATE_STREAM) {*ppacket = packet = new SrsCreateStreamPacket();return packet->decode(stream);} else if (command == RTMP_AMF0_COMMAND_PLAY) {*ppacket = packet = new SrsPlayPacket(); //RTMP拉流return packet->decode(stream);} else if (command == RTMP_AMF0_COMMAND_PAUSE) {*ppacket = packet = new SrsPausePacket();return packet->decode(stream);} else if (command == RTMP_AMF0_COMMAND_RELEASE_STREAM) {*ppacket = packet = new SrsFMLEStartPacket(); //RTMP推流return packet->decode(stream);} else if (command == RTMP_AMF0_COMMAND_FC_PUBLISH) {....return err;
}

推流时解析协议控制消息调用栈如下，拉流调用栈也是一样。

(gdb) bt
#0  SrsFMLEStartPacket::SrsFMLEStartPacket (this=0x105beb0) at src/protocol/srs_rtmp_stack.cpp:3556
#1  0x0000000000483841 in SrsProtocol::do_decode_message (this=0x1040110, header=..., stream=0x105a250, ppacket=0x105a240) at src/protocol/srs_rtmp_stack.cpp:700
#2  0x00000000004826ed in SrsProtocol::decode_message (this=0x1040110, msg=0x105bc40, ppacket=0x105a2c8) at src/protocol/srs_rtmp_stack.cpp:413
#3  0x000000000048d851 in SrsRtmpServer::identify_client (this=0x10400e0, stream_id=1, type=@0x1049998: SrsRtmpConnUnknown, stream_name="", duration=@0x1049b18: -1)at src/protocol/srs_rtmp_stack.cpp:2510
#4  0x000000000050090c in SrsRtmpConn::stream_service_cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:454
#5  0x0000000000500512 in SrsRtmpConn::service_cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:403
#6  0x00000000004fefd1 in SrsRtmpConn::do_cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:216
#7  0x0000000000507f59 in SrsRtmpConn::cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:1457
#8  0x00000000005379d8 in SrsFastCoroutine::cycle (this=0x1040040) at src/app/srs_app_st.cpp:272
#9  0x0000000000537a6e in SrsFastCoroutine::pfn (arg=0x1040040) at src/app/srs_app_st.cpp:287
#10 0x000000000065010b in _st_thread_main () at sched.c:363
#11 0x0000000000650988 in st_thread_create (start=0xf78f00, arg=0x101040040, joinable=0, stk_size=6640311) at sched.c:694
Backtrace stopped: previous frame inner to this frame (corrupt stack?)

当SrsProtocol::decode_message函数中返回的对象是SrsFMLEStartPacket，会调用SrsRtmpServer::identify_fmle_publish_client，type赋值为SrsRtmpConnFMLEPublish，表示推流。

srs_error_t SrsRtmpServer::identify_fmle_publish_client(SrsFMLEStartPacket* req, SrsRtmpConnType& type, string& stream_name)
{srs_error_t err = srs_success;type = SrsRtmpConnFMLEPublish;stream_name = req->stream_name;// releaseStream responseif (true) {SrsFMLEStartResPacket* pkt = new SrsFMLEStartResPacket(req->transaction_id);if ((err = protocol->send_and_free_packet(pkt, 0)) != srs_success) {return srs_error_wrap(err, "send releaseStream response");}}return err;
}

当SrsProtocol::decode_message函数中返回的对象是SrsPlayPacket，会调用SrsRtmpServer::identify_play_client，type赋值为SrsRtmpConnPlay，表示拉流。

srs_error_t SrsRtmpServer::identify_play_client(SrsPlayPacket* req, SrsRtmpConnType& type, string& stream_name, srs_utime_t& duration)
{type = SrsRtmpConnPlay;stream_name = req->stream_name;duration = srs_utime_t(req->duration) * SRS_UTIME_MILLISECONDS;return srs_success;
}

获取SrsLiveSource，如果没有则创建。推流时创建，一个推流对应一个source，同一个source对应多个拉流。

srs_error_t SrsLiveSourceManager::fetch_or_create(SrsRequest* r, ISrsLiveSourceHandler* h, SrsLiveSource** pps)
{srs_error_t err = srs_success;// Use lock to protect coroutine switch.// @bug https://github.com/ossrs/srs/issues/1230// TODO: FIXME: Use smaller lock.SrsLocker(lock);SrsLiveSource* source = NULL;if ((source = fetch(r)) != NULL) { //通过stream_url在SrsLiveSourceManager的pool（map）中查询对应的SrsLiveSource// we always update the request of resource,// for origin auth is on, the token in request maybe invalid,// and we only need to update the token of request, it's simple.source->update_auth(r);*pps = source;return err;}string stream_url = r->get_stream_url();string vhost = r->vhost;// should always not exists for create a source.srs_assert (pool.find(stream_url) == pool.end());srs_trace("new source, stream_url=%s", stream_url.c_str());source = new SrsLiveSource(); //没找到则创建并进行初始化if ((err = source->initialize(r, h)) != srs_success) {err = srs_error_wrap(err, "init source %s", r->get_stream_url().c_str());goto failed;}pool[stream_url] = source;*pps = source;return err;failed:srs_freep(source);return err;
}

2. 启动推流

推流最终会通过SrsRtmpConn::publishing函数调用SrsRtmpConn::do_publishing中，调用栈为：

(gdb) bt
#0  SrsRtmpConn::do_publishing (this=0x103fcc0, source=0x105c250, rtrd=0x105a280) at src/app/srs_app_rtmp_conn.cpp:857
#1  0x000000000050432a in SrsRtmpConn::publishing (this=0x103fcc0, source=0x105c250) at src/app/srs_app_rtmp_conn.cpp:838
#2  0x00000000005016dd in SrsRtmpConn::stream_service_cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:542
#3  0x0000000000500512 in SrsRtmpConn::service_cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:403
#4  0x00000000004fefd1 in SrsRtmpConn::do_cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:216
#5  0x0000000000507f59 in SrsRtmpConn::cycle (this=0x103fcc0) at src/app/srs_app_rtmp_conn.cpp:1457
#6  0x00000000005379d8 in SrsFastCoroutine::cycle (this=0x1040040) at src/app/srs_app_st.cpp:272
#7  0x0000000000537a6e in SrsFastCoroutine::pfn (arg=0x1040040) at src/app/srs_app_st.cpp:287
#8  0x000000000065010b in _st_thread_main () at sched.c:363
#9  0x0000000000650988 in st_thread_create (start=0xf78f00, arg=0x101040040, joinable=0, stk_size=6640311) at sched.c:694
Backtrace stopped: previous frame inner to this frame (corrupt stack?)

SrsRtmpConn::publishing会创建SrsPublishRecvThread协程来接收数据，然后在SrsRtmpConn::do_publishing开启接收协程。

srs_error_t SrsRtmpConn::publishing(SrsLiveSource* source)
{srs_error_t err = srs_success;SrsRequest* req = info->req;if (_srs_config->get_refer_enabled(req->vhost)) {if ((err = refer->check(req->pageUrl, _srs_config->get_refer_publish(req->vhost))) != srs_success) {return srs_error_wrap(err, "rtmp: referer check");}}if ((err = http_hooks_on_publish()) != srs_success) {return srs_error_wrap(err, "rtmp: callback on publish");}// TODO: FIXME: Should refine the state of publishing.if ((err = acquire_publish(source)) == srs_success) {// use isolate thread to recv,// @see: https://github.com/ossrs/srs/issues/237SrsPublishRecvThread rtrd(rtmp, req, srs_netfd_fileno(stfd), 0, this, source, _srs_context->get_id());err = do_publishing(source, &rtrd);rtrd.stop();}// whatever the acquire publish, always release publish.// when the acquire error in the midlle-way, the publish state changed,// but failed, so we must cleanup it.// @see https://github.com/ossrs/srs/issues/474// @remark when stream is busy, should never release it.if (srs_error_code(err) != ERROR_SYSTEM_STREAM_BUSY) {release_publish(source);}http_hooks_on_unpublish();return err;
}

srs_error_t SrsRtmpConn::do_publishing(SrsLiveSource* source, SrsPublishRecvThread* rtrd)
{srs_error_t err = srs_success;SrsRequest* req = info->req;SrsPithyPrint* pprint = SrsPithyPrint::create_rtmp_publish();SrsAutoFree(SrsPithyPrint, pprint);// update the statistic when source disconveried.SrsStatistic* stat = SrsStatistic::instance();if ((err = stat->on_client(_srs_context->get_id().c_str(), req, this, info->type)) != srs_success) {return srs_error_wrap(err, "rtmp: stat client");}// start isolate recv thread.// TODO: FIXME: Pass the callback here.if ((err = rtrd->start()) != srs_success) {return srs_error_wrap(err, "rtmp: receive thread");}...return err;
}

最终会执行到SrsRecvThread::do_cycle，具体逻辑见第三部分：服务器读取RTMP推流数据

3. 启动拉流

拉流最终会通过SrsRtmpConn::playing函数调用SrsRtmpConn::do_playing中，调用栈为：

(gdb) bt
#0  SrsRtmpConn::do_playing (this=0x10acda0, source=0x105c280, consumer=0x10c8b60, rtrd=0x10c7230) at src/app/srs_app_rtmp_conn.cpp:688
#1  0x0000000000502ccb in SrsRtmpConn::playing (this=0x10acda0, source=0x105c280) at src/app/srs_app_rtmp_conn.cpp:674
#2  0x0000000000501600 in SrsRtmpConn::stream_service_cycle (this=0x10acda0) at src/app/srs_app_rtmp_conn.cpp:532
#3  0x0000000000500512 in SrsRtmpConn::service_cycle (this=0x10acda0) at src/app/srs_app_rtmp_conn.cpp:403
#4  0x00000000004fefd1 in SrsRtmpConn::do_cycle (this=0x10acda0) at src/app/srs_app_rtmp_conn.cpp:216
#5  0x0000000000507f59 in SrsRtmpConn::cycle (this=0x10acda0) at src/app/srs_app_rtmp_conn.cpp:1457
#6  0x00000000005379d8 in SrsFastCoroutine::cycle (this=0x10ad050) at src/app/srs_app_st.cpp:272
#7  0x0000000000537a6e in SrsFastCoroutine::pfn (arg=0x10ad050) at src/app/srs_app_st.cpp:287
#8  0x000000000065010b in _st_thread_main () at sched.c:363
#9  0x0000000000650988 in st_thread_create (start=0xf78f00, arg=0x1010ad050, joinable=0, stk_size=6640311) at sched.c:694
Backtrace stopped: previous frame inner to this frame (corrupt stack?)

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SRS流媒体服务器——RTMP推流、拉流创建连接

SRS流媒体服务器——RTMP推流、拉流创建连接

目录

1. RTMP推流、拉流创建连接

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

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