srs/trunk/src/app/srs_app_rtmp_conn.cpp
2025-09-27 09:40:57 -04:00

1609 lines
52 KiB
C++

//
// Copyright (c) 2013-2025 The SRS Authors
//
// SPDX-License-Identifier: MIT
//
#include <srs_app_rtmp_conn.hpp>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <netinet/tcp.h>
#include <stdlib.h>
#include <sys/socket.h>
#include <unistd.h>
using namespace std;
#include <srs_app_config.hpp>
#include <srs_app_edge.hpp>
#include <srs_app_hls.hpp>
#include <srs_app_http_hooks.hpp>
#include <srs_app_recv_thread.hpp>
#include <srs_app_refer.hpp>
#include <srs_app_rtc_source.hpp>
#include <srs_app_rtmp_source.hpp>
#include <srs_app_security.hpp>
#include <srs_app_server.hpp>
#include <srs_app_srt_source.hpp>
#include <srs_app_st.hpp>
#include <srs_app_statistic.hpp>
#include <srs_app_stream_token.hpp>
#include <srs_kernel_pithy_print.hpp>
#include <srs_app_utility.hpp>
#include <srs_core_autofree.hpp>
#include <srs_core_performance.hpp>
#include <srs_kernel_error.hpp>
#include <srs_kernel_log.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_protocol_amf0.hpp>
#include <srs_protocol_json.hpp>
#include <srs_protocol_rtmp_msg_array.hpp>
#include <srs_protocol_rtmp_stack.hpp>
#include <srs_protocol_utility.hpp>
#ifdef SRS_RTSP
#include <srs_app_rtsp_source.hpp>
#endif
// the timeout in srs_utime_t to wait encoder to republish
// if timeout, close the connection.
#define SRS_REPUBLISH_SEND_TIMEOUT (3 * SRS_UTIME_MINUTES)
// if timeout, close the connection.
#define SRS_REPUBLISH_RECV_TIMEOUT (3 * SRS_UTIME_MINUTES)
// the timeout in srs_utime_t to wait client data, when client paused
// if timeout, close the connection.
#define SRS_PAUSED_SEND_TIMEOUT (3 * SRS_UTIME_MINUTES)
// if timeout, close the connection.
#define SRS_PAUSED_RECV_TIMEOUT (3 * SRS_UTIME_MINUTES)
// when edge timeout, retry next.
#define SRS_EDGE_TOKEN_TRAVERSE_TIMEOUT (3 * SRS_UTIME_SECONDS)
SrsSimpleRtmpClient::SrsSimpleRtmpClient(string u, srs_utime_t ctm, srs_utime_t stm) : SrsBasicRtmpClient(u, ctm, stm)
{
}
SrsSimpleRtmpClient::~SrsSimpleRtmpClient()
{
}
srs_error_t SrsSimpleRtmpClient::connect_app()
{
std::vector<SrsIPAddress *> &ips = srs_get_local_ips();
srs_assert(_srs_config->get_stats_network() < (int)ips.size());
SrsIPAddress *local_ip = ips[_srs_config->get_stats_network()];
bool debug_srs_upnode = _srs_config->get_debug_srs_upnode(req_->vhost_);
return do_connect_app(local_ip->ip_, debug_srs_upnode);
}
SrsClientInfo::SrsClientInfo()
{
edge_ = false;
req_ = new SrsRequest();
res_ = new SrsResponse();
type_ = SrsRtmpConnUnknown;
}
SrsClientInfo::~SrsClientInfo()
{
srs_freep(req_);
srs_freep(res_);
}
SrsRtmpTransport::SrsRtmpTransport(srs_netfd_t c)
{
stfd_ = c;
skt_ = new SrsTcpConnection(c);
}
SrsRtmpTransport::~SrsRtmpTransport()
{
srs_freep(skt_);
}
srs_netfd_t SrsRtmpTransport::fd()
{
return stfd_;
}
ISrsProtocolReadWriter *SrsRtmpTransport::io()
{
return skt_;
}
srs_error_t SrsRtmpTransport::handshake()
{
return srs_success;
}
const char *SrsRtmpTransport::transport_type()
{
return "plaintext";
}
srs_error_t SrsRtmpTransport::set_socket_buffer(srs_utime_t buffer_v)
{
return skt_->set_socket_buffer(buffer_v);
}
srs_error_t SrsRtmpTransport::set_tcp_nodelay(bool v)
{
return skt_->set_tcp_nodelay(v);
}
int64_t SrsRtmpTransport::get_recv_bytes()
{
return skt_->get_recv_bytes();
}
int64_t SrsRtmpTransport::get_send_bytes()
{
return skt_->get_send_bytes();
}
SrsRtmpsTransport::SrsRtmpsTransport(srs_netfd_t c) : SrsRtmpTransport(c)
{
ssl_ = new SrsSslConnection(skt_);
}
SrsRtmpsTransport::~SrsRtmpsTransport()
{
srs_freep(ssl_);
}
ISrsProtocolReadWriter *SrsRtmpsTransport::io()
{
return ssl_;
}
srs_error_t SrsRtmpsTransport::handshake()
{
string crt_file = _srs_config->get_rtmps_ssl_cert();
string key_file = _srs_config->get_rtmps_ssl_key();
srs_error_t err = ssl_->handshake(key_file, crt_file);
if (err != srs_success) {
return srs_error_wrap(err, "ssl handshake");
}
return srs_success;
}
const char *SrsRtmpsTransport::transport_type()
{
return "ssl";
}
SrsRtmpConn::SrsRtmpConn(SrsServer *svr, SrsRtmpTransport *transport, string cip, int cport)
{
// Create a identify for this client.
_srs_context->set_id(_srs_context->generate_id());
server_ = svr;
transport_ = transport;
manager_ = _srs_conn_manager;
ip_ = cip;
port_ = cport;
create_time_ = srsu2ms(srs_time_now_cached());
trd_ = new SrsSTCoroutine("rtmp", this, _srs_context->get_id());
kbps_ = new SrsNetworkKbps();
kbps_->set_io(transport_->io(), transport_->io());
delta_ = new SrsNetworkDelta();
delta_->set_io(transport_->io(), transport_->io());
rtmp_ = new SrsRtmpServer(transport_->io());
refer_ = new SrsRefer();
security_ = new SrsSecurity();
duration_ = 0;
wakable_ = NULL;
mw_sleep_ = SRS_PERF_MW_SLEEP;
mw_msgs_ = 0;
realtime_ = SRS_PERF_MIN_LATENCY_ENABLED;
send_min_interval_ = 0;
tcp_nodelay_ = false;
info_ = new SrsClientInfo();
publish_1stpkt_timeout_ = 0;
publish_normal_timeout_ = 0;
_srs_config->subscribe(this);
}
SrsRtmpConn::~SrsRtmpConn()
{
_srs_config->unsubscribe(this);
trd_->interrupt();
// wakeup the handler which need to notice.
if (wakable_) {
wakable_->wakeup();
}
srs_freep(trd_);
srs_freep(kbps_);
srs_freep(delta_);
srs_freep(transport_);
srs_freep(info_);
srs_freep(rtmp_);
srs_freep(refer_);
srs_freep(security_);
#ifdef SRS_APM
srs_freep(span_main_);
srs_freep(span_connect_);
srs_freep(span_client_);
#endif
}
std::string SrsRtmpConn::desc()
{
return "RtmpConn";
}
std::string srs_ipv4_string(uint32_t rip)
{
return srs_fmt_sprintf("%d.%d.%d.%d", uint8_t(rip >> 24), uint8_t(rip >> 16), uint8_t(rip >> 8), uint8_t(rip));
}
// TODO: return detail message when error for client.
srs_error_t SrsRtmpConn::do_cycle()
{
srs_error_t err = srs_success;
#ifdef SRS_APM
srs_trace("RTMP client transport=%s, ip=%s:%d, fd=%d, trace=%s, span=%s", transport_->transport_type(), ip.c_str(), port, srs_netfd_fileno(transport_->fd()),
span_main_->format_trace_id(), span_main_->format_span_id());
#else
srs_trace("RTMP client transport=%s, ip=%s:%d, fd=%d", transport_->transport_type(), ip_.c_str(), port_, srs_netfd_fileno(transport_->fd()));
#endif
if ((err = transport_->handshake()) != srs_success) {
return srs_error_wrap(err, "transport handshake");
}
rtmp_->set_recv_timeout(SRS_CONSTS_RTMP_TIMEOUT);
rtmp_->set_send_timeout(SRS_CONSTS_RTMP_TIMEOUT);
if ((err = rtmp_->handshake()) != srs_success) {
return srs_error_wrap(err, "rtmp handshake");
}
uint32_t rip = rtmp_->proxy_real_ip();
std::string rips = srs_ipv4_string(rip);
if (rip > 0) {
srs_trace("RTMP proxy real client ip=%s", rips.c_str());
}
ISrsRequest *req = info_->req_;
if ((err = rtmp_->connect_app(req)) != srs_success) {
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"));
// show client identity
if (req->args_) {
std::string srs_version;
std::string srs_server_ip;
int srs_pid = 0;
int srs_id = 0;
SrsAmf0Any *prop = NULL;
if ((prop = req->args_->ensure_property_string("srs_version")) != NULL) {
srs_version = prop->to_str();
}
if ((prop = req->args_->ensure_property_string("srs_server_ip")) != NULL) {
srs_server_ip = prop->to_str();
}
if ((prop = req->args_->ensure_property_number("srs_pid")) != NULL) {
srs_pid = (int)prop->to_number();
}
if ((prop = req->args_->ensure_property_number("srs_id")) != NULL) {
srs_id = (int)prop->to_number();
}
if (srs_pid > 0) {
srs_trace("edge-srs ip=%s, version=%s, pid=%d, id=%d",
srs_server_ip.c_str(), srs_version.c_str(), srs_pid, srs_id);
}
}
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_freep(err);
}
return err;
}
ISrsKbpsDelta *SrsRtmpConn::delta()
{
return delta_;
}
srs_error_t SrsRtmpConn::service_cycle()
{
srs_error_t err = srs_success;
ISrsRequest *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), SrsPeerBandwidthDynamic)) != 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(transport_->fd()));
// 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.
if ((err = rtmp_->response_connect_app(req, local_ip.c_str())) != srs_success) {
return srs_error_wrap(err, "rtmp: response connect app");
}
#ifdef SRS_APM
// Must be a connecting application span.
span_connect_->end();
#endif
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;
}
// when not system control error, fatal error, return.
if (!srs_is_system_control_error(err)) {
return srs_error_wrap(err, "rtmp: stream service");
}
// for republish, continue service
if (srs_error_code(err) == ERROR_CONTROL_REPUBLISH) {
// set timeout to a larger value, wait for encoder to republish.
rtmp_->set_send_timeout(SRS_REPUBLISH_RECV_TIMEOUT);
rtmp_->set_recv_timeout(SRS_REPUBLISH_SEND_TIMEOUT);
srs_info("rtmp: retry for republish");
srs_freep(err);
continue;
}
// for "some" system control error,
// logical accept and retry stream service.
if (srs_error_code(err) == ERROR_CONTROL_RTMP_CLOSE) {
// TODO: FIXME: use ping message to anti-death of socket.
// set timeout to a larger value, for user paused.
rtmp_->set_recv_timeout(SRS_PAUSED_RECV_TIMEOUT);
rtmp_->set_send_timeout(SRS_PAUSED_SEND_TIMEOUT);
srs_trace("rtmp: retry for close");
srs_freep(err);
continue;
}
// for other system control message, fatal error.
return srs_error_wrap(err, "rtmp: reject");
}
return err;
}
srs_error_t SrsRtmpConn::stream_service_cycle()
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
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_net_url_parse_tcurl(req->tcUrl_, req->schema_, req->host_, req->vhost_, req->app_, req->stream_, req->port_, req->param_);
// guess stream name
if (req->stream_.empty()) {
string app = req->app_, param = req->param_;
srs_net_url_guess_stream(req->app_, req->param_, req->stream_);
srs_trace("Guessing by app=%s, param=%s to app=%s, param=%s, stream=%s", app.c_str(), param.c_str(), req->app_.c_str(), req->param_.c_str(), req->stream_.c_str());
}
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 config
SrsConfDirective *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/239
if (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 check
if ((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/834
if (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);
// Acquire stream publish token to prevent race conditions across all protocols.
SrsStreamPublishToken *publish_token_raw = NULL;
if (info_->type_ != SrsRtmpConnPlay && (err = _srs_stream_publish_tokens->acquire_token(req, publish_token_raw)) != srs_success) {
return srs_error_wrap(err, "acquire stream publish token");
}
SrsUniquePtr<SrsStreamPublishToken> publish_token(publish_token_raw);
if (publish_token.get()) {
srs_trace("stream publish token acquired, type=%s, url=%s",
srs_client_type_string(info_->type_).c_str(), req->get_stream_url().c_str());
}
// find a source to serve.
SrsSharedPtr<SrsLiveSource> live_source;
if ((err = _srs_sources->fetch_or_create(req, live_source)) != srs_success) {
return srs_error_wrap(err, "rtmp: fetch source");
}
srs_assert(live_source.get() != NULL);
bool enabled_cache = _srs_config->get_gop_cache(req->vhost_);
int gcmf = _srs_config->get_gop_cache_max_frames(req->vhost_);
srs_trace("source url=%s, ip=%s, cache=%d/%d, is_edge=%d, source_id=%s/%s",
req->get_stream_url().c_str(), ip_.c_str(), enabled_cache, gcmf, info_->edge_, live_source->source_id().c_str(),
live_source->pre_source_id().c_str());
live_source->set_cache(enabled_cache);
live_source->set_gop_cache_max_frames(gcmf);
switch (info_->type_) {
case SrsRtmpConnPlay: {
// response connection start play
if ((err = rtmp_->start_play(info_->res_->stream_id_)) != srs_success) {
return srs_error_wrap(err, "rtmp: start play");
}
// We must do stat the client before hooks, because hooks depends on it.
SrsStatistic *stat = _srs_stat;
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");
}
// We must do hook after stat, because depends on it.
if ((err = http_hooks_on_play()) != srs_success) {
return srs_error_wrap(err, "rtmp: callback on play");
}
#ifdef SRS_APM
// Must be a client span.
span_client_->set_name("play")->end();
// We end the connection span because it's a producer and only trace the established.
span_main_->end();
#endif
err = playing(live_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");
}
#ifdef SRS_APM
// Must be a client span.
span_client_->set_name("publish")->end();
// We end the connection span because it's a producer and only trace the established.
span_main_->end();
#endif
return publishing(live_source);
}
case SrsRtmpConnHaivisionPublish: {
if ((err = rtmp_->start_haivision_publish(info_->res_->stream_id_)) != srs_success) {
return srs_error_wrap(err, "rtmp: start HAIVISION publish");
}
#ifdef SRS_APM
// Must be a client span.
span_client_->set_name("publish")->end();
// We end the connection span because it's a producer and only trace the established.
span_main_->end();
#endif
return publishing(live_source);
}
case SrsRtmpConnFlashPublish: {
if ((err = rtmp_->start_flash_publish(info_->res_->stream_id_)) != srs_success) {
return srs_error_wrap(err, "rtmp: start FLASH publish");
}
#ifdef SRS_APM
// Must be a client span.
span_client_->set_name("publish")->end();
// We end the connection span because it's a producer and only trace the established.
span_main_->end();
#endif
return publishing(live_source);
}
default: {
return srs_error_new(ERROR_SYSTEM_CLIENT_INVALID, "rtmp: unknown client type=%d", info_->type_);
}
}
return err;
}
srs_error_t SrsRtmpConn::check_vhost(bool try_default_vhost)
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
srs_assert(req != NULL);
SrsConfDirective *vhost = _srs_config->get_vhost(req->vhost_, try_default_vhost);
if (vhost == NULL) {
return srs_error_new(ERROR_RTMP_VHOST_NOT_FOUND, "rtmp: no vhost %s", req->vhost_.c_str());
}
if (!_srs_config->get_vhost_enabled(req->vhost_)) {
return srs_error_new(ERROR_RTMP_VHOST_NOT_FOUND, "rtmp: vhost %s disabled", req->vhost_.c_str());
}
if (req->vhost_ != vhost->arg0()) {
srs_trace("vhost change from %s to %s", req->vhost_.c_str(), vhost->arg0().c_str());
req->vhost_ = vhost->arg0();
}
if (_srs_config->get_refer_enabled(req->vhost_)) {
if ((err = refer_->check(req->pageUrl_, _srs_config->get_refer_all(req->vhost_))) != srs_success) {
return srs_error_wrap(err, "rtmp: referer check");
}
}
if ((err = http_hooks_on_connect()) != srs_success) {
return srs_error_wrap(err, "rtmp: callback on connect");
}
return err;
}
srs_error_t SrsRtmpConn::playing(SrsSharedPtr<SrsLiveSource> source)
{
srs_error_t err = srs_success;
// Check page referer of player.
ISrsRequest *req = info_->req_;
if (_srs_config->get_refer_enabled(req->vhost_)) {
if ((err = refer_->check(req->pageUrl_, _srs_config->get_refer_play(req->vhost_))) != srs_success) {
return srs_error_wrap(err, "rtmp: referer check");
}
}
// When origin cluster enabled, try to redirect to the origin which is active.
// A active origin is a server which is delivering stream.
if (!info_->edge_ && _srs_config->get_vhost_origin_cluster(req->vhost_) && source->inactive()) {
vector<string> coworkers = _srs_config->get_vhost_coworkers(req->vhost_);
for (int i = 0; i < (int)coworkers.size(); i++) {
// TODO: FIXME: User may config the server itself as coworker, we must identify and ignore it.
string host;
int port = 0;
string coworker = coworkers.at(i);
string url = "http://" + coworker + "/api/v1/clusters?" + "vhost=" + req->vhost_ + "&ip=" + req->host_ + "&app=" + req->app_ + "&stream=" + req->stream_ + "&coworker=" + coworker;
if ((err = _srs_hooks->discover_co_workers(url, host, port)) != srs_success) {
// If failed to discovery stream in this coworker, we should request the next one util the last.
// @see https://github.com/ossrs/srs/issues/1223
if (i < (int)coworkers.size() - 1) {
continue;
}
return srs_error_wrap(err, "discover coworkers, url=%s", url.c_str());
}
string rurl = srs_net_url_encode_rtmp_url(host, port, req->host_, req->vhost_, req->app_, req->stream_, req->param_);
srs_trace("rtmp: redirect in cluster, from=%s:%d, target=%s:%d, url=%s, rurl=%s",
req->host_.c_str(), req->port_, host.c_str(), port, url.c_str(), rurl.c_str());
// Ignore if host or port is invalid.
if (host.empty() || port == 0) {
continue;
}
bool accepted = false;
if ((err = rtmp_->redirect(req, rurl, accepted)) != srs_success) {
srs_freep(err);
} else {
return srs_error_new(ERROR_CONTROL_REDIRECT, "redirected");
}
}
return srs_error_new(ERROR_OCLUSTER_REDIRECT, "no origin");
}
// Set the socket options for transport.
set_sock_options();
// Create a consumer of source.
SrsLiveConsumer *consumer_raw = NULL;
if ((err = source->create_consumer(consumer_raw)) != srs_success) {
return srs_error_wrap(err, "rtmp: create consumer");
}
SrsUniquePtr<SrsLiveConsumer> consumer(consumer_raw);
if ((err = source->consumer_dumps(consumer.get())) != srs_success) {
return srs_error_wrap(err, "rtmp: dumps consumer");
}
// Use receiving thread to receive packets from peer.
SrsQueueRecvThread trd(consumer.get(), rtmp_, SRS_PERF_MW_SLEEP, _srs_context->get_id());
if ((err = trd.start()) != srs_success) {
return srs_error_wrap(err, "rtmp: start receive thread");
}
// Deliver packets to peer.
wakable_ = consumer.get();
err = do_playing(source, consumer.get(), &trd);
wakable_ = NULL;
trd.stop();
// Drop all packets in receiving thread.
if (!trd.empty()) {
srs_warn("drop the received %d messages", trd.size());
}
return err;
}
srs_error_t SrsRtmpConn::do_playing(SrsSharedPtr<SrsLiveSource> source, SrsLiveConsumer *consumer, SrsQueueRecvThread *rtrd)
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
srs_assert(req);
srs_assert(consumer);
// initialize other components
SrsUniquePtr<SrsPithyPrint> pprint(SrsPithyPrint::create_rtmp_play());
SrsMessageArray msgs(SRS_PERF_MW_MSGS);
bool user_specified_duration_to_stop = (req->duration_ > 0);
int64_t starttime = -1;
// setup the realtime.
realtime_ = _srs_config->get_realtime_enabled(req->vhost_, false);
// setup the mw config.
// when mw_sleep changed, resize the socket send buffer.
mw_msgs_ = _srs_config->get_mw_msgs(req->vhost_, realtime_, false);
mw_sleep_ = _srs_config->get_mw_sleep(req->vhost_);
transport_->set_socket_buffer(mw_sleep_);
// initialize the send_min_interval
send_min_interval_ = _srs_config->get_send_min_interval(req->vhost_);
srs_trace("start play smi=%dms, mw_sleep=%d, mw_msgs=%d, realtime=%d, tcp_nodelay=%d",
srsu2msi(send_min_interval_), srsu2msi(mw_sleep_), mw_msgs_, realtime_, tcp_nodelay_);
while (true) {
// when source is set to expired, disconnect it.
if ((err = trd_->pull()) != srs_success) {
return srs_error_wrap(err, "rtmp: thread quit");
}
// collect elapse for pithy print.
pprint->elapse();
// to use isolate thread to recv, can improve about 33% performance.
while (!rtrd->empty()) {
SrsRtmpCommonMessage *msg = rtrd->pump();
if ((err = process_play_control_msg(consumer, msg)) != srs_success) {
return srs_error_wrap(err, "rtmp: play control message");
}
}
// quit when recv thread error.
if ((err = rtrd->error_code()) != srs_success) {
return srs_error_wrap(err, "rtmp: recv thread");
}
#ifdef SRS_PERF_QUEUE_COND_WAIT
// wait for message to incoming.
// @see https://github.com/ossrs/srs/issues/257
consumer->wait(mw_msgs_, mw_sleep_);
#endif
// get messages from consumer.
// each msg in msgs.msgs must be free, for the SrsMessageArray never free them.
// @remark when enable send_min_interval, only fetch one message a time.
int count = (send_min_interval_ > 0) ? 1 : 0;
if ((err = consumer->dump_packets(&msgs, count)) != srs_success) {
return srs_error_wrap(err, "rtmp: consumer dump packets");
}
// reportable
if (pprint->can_print()) {
kbps_->sample();
srs_trace("-> " SRS_CONSTS_LOG_PLAY " time=%d, msgs=%d, okbps=%d,%d,%d, ikbps=%d,%d,%d, mw=%d/%d",
(int)pprint->age(), count, kbps_->get_send_kbps(), kbps_->get_send_kbps_30s(), kbps_->get_send_kbps_5m(),
kbps_->get_recv_kbps(), kbps_->get_recv_kbps_30s(), kbps_->get_recv_kbps_5m(), srsu2msi(mw_sleep_), mw_msgs_);
#ifdef SRS_APM
// TODO: Do not use pithy print for frame span.
ISrsApmSpan *sample = _srs_apm->span("play-frame")->set_kind(SrsApmKindConsumer)->as_child(span.get())->attr("msgs", srs_fmt_sprintf("%d", count))->attr("kbps", srs_fmt_sprintf("%d", kbps_->get_send_kbps_30s()));
srs_freep(sample);
#endif
}
if (count <= 0) {
#ifndef SRS_PERF_QUEUE_COND_WAIT
srs_usleep(mw_sleep);
#endif
// ignore when nothing got.
continue;
}
// only when user specifies the duration,
// we start to collect the durations for each message.
if (user_specified_duration_to_stop) {
for (int i = 0; i < count; i++) {
SrsMediaPacket *msg = msgs.msgs_[i];
// foreach msg, collect the duration.
// @remark: never use msg when sent it, for the protocol sdk will free it.
if (starttime < 0 || starttime > msg->timestamp_) {
starttime = msg->timestamp_;
}
duration_ += (msg->timestamp_ - starttime) * SRS_UTIME_MILLISECONDS;
starttime = msg->timestamp_;
}
}
// sendout messages, all messages are freed by send_and_free_messages().
// no need to assert msg, for the rtmp will assert it.
if (count > 0 && (err = rtmp_->send_and_free_messages(msgs.msgs_, count, info_->res_->stream_id_)) != srs_success) {
return srs_error_wrap(err, "rtmp: send %d messages", count);
}
// if duration specified, and exceed it, stop play live.
// @see: https://github.com/ossrs/srs/issues/45
if (user_specified_duration_to_stop) {
if (duration_ >= req->duration_) {
return srs_error_new(ERROR_RTMP_DURATION_EXCEED, "rtmp: time %d up %d", srsu2msi(duration_), srsu2msi(req->duration_));
}
}
// apply the minimal interval for delivery stream in srs_utime_t.
if (send_min_interval_ > 0) {
srs_usleep(send_min_interval_);
}
// Yield to another coroutines.
// @see https://github.com/ossrs/srs/issues/2194#issuecomment-777437476
srs_thread_yield();
}
return err;
}
srs_error_t SrsRtmpConn::publishing(SrsSharedPtr<SrsLiveSource> source)
{
srs_error_t err = srs_success;
ISrsRequest *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");
}
}
// We must do stat the client before hooks, because hooks depends on it.
SrsStatistic *stat = _srs_stat;
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");
}
// We must do hook after stat, because depends on it.
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.
srs_error_t acquire_err = acquire_publish(source);
if ((err = acquire_err) == srs_success) {
// use isolate thread to recv,
// @see: https://github.com/ossrs/srs/issues/237
SrsPublishRecvThread rtrd(rtmp_, req, srs_netfd_fileno(transport_->fd()), 0, this, source, _srs_context->get_id());
err = do_publishing(source, &rtrd);
rtrd.stop();
}
// Release and callback when acquire publishing success, if not, we should ignore, because the source
// is not published by this session.
if (acquire_err == srs_success) {
release_publish(source);
http_hooks_on_unpublish();
}
return err;
}
srs_error_t SrsRtmpConn::do_publishing(SrsSharedPtr<SrsLiveSource> source, SrsPublishRecvThread *rtrd)
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
SrsUniquePtr<SrsPithyPrint> pprint(SrsPithyPrint::create_rtmp_publish());
// start isolate recv thread.
// TODO: FIXME: Pass the callback here.
if ((err = rtrd->start()) != srs_success) {
return srs_error_wrap(err, "rtmp: receive thread");
}
// initialize the publish timeout.
publish_1stpkt_timeout_ = _srs_config->get_publish_1stpkt_timeout(req->vhost_);
publish_normal_timeout_ = _srs_config->get_publish_normal_timeout(req->vhost_);
srs_utime_t publish_kickoff_for_idle = _srs_config->get_publish_kickoff_for_idle(req->vhost_);
// set the sock options.
set_sock_options();
if (true) {
bool mr = _srs_config->get_mr_enabled(req->vhost_);
srs_utime_t mr_sleep = _srs_config->get_mr_sleep(req->vhost_);
srs_trace("start publish mr=%d/%d, p1stpt=%d, pnt=%d, tcp_nodelay=%d", mr, srsu2msi(mr_sleep), srsu2msi(publish_1stpkt_timeout_), srsu2msi(publish_normal_timeout_), tcp_nodelay_);
}
#ifdef SRS_APM
SrsUniquePtr<ISrsApmSpan> span(_srs_apm->span("publish-cycle")->set_kind(SrsApmKindProducer)->as_child(span_client_)->attr("timeout", srs_fmt_sprintf("%d", srsu2msi(publish_normal_timeout_)))->end());
#endif
// Response the start publishing message, let client start to publish messages.
if ((err = rtmp_->start_publishing(info_->res_->stream_id_)) != srs_success) {
return srs_error_wrap(err, "start publishing");
}
int64_t nb_msgs = 0;
uint64_t nb_frames = 0;
while (true) {
if ((err = trd_->pull()) != srs_success) {
return srs_error_wrap(err, "rtmp: thread quit");
}
// Kick off the publisher when idle for a period of timeout.
if (source->publisher_is_idle_for(publish_kickoff_for_idle)) {
return srs_error_new(ERROR_KICKOFF_FOR_IDLE, "kicked for idle, url=%s, timeout=%ds", req->tcUrl_.c_str(), srsu2si(publish_kickoff_for_idle));
}
pprint->elapse();
// cond wait for timeout.
if (nb_msgs == 0) {
// when not got msgs, wait for a larger timeout.
rtrd->wait(publish_1stpkt_timeout_);
} else {
rtrd->wait(publish_normal_timeout_);
}
// check the thread error code.
if ((err = rtrd->error_code()) != srs_success) {
return srs_error_wrap(err, "rtmp: receive thread");
}
// when not got any messages, timeout.
if (rtrd->nb_msgs() <= nb_msgs) {
return srs_error_new(ERROR_SOCKET_TIMEOUT, "rtmp: publish timeout %dms, nb_msgs=%d",
nb_msgs ? srsu2msi(publish_normal_timeout_) : srsu2msi(publish_1stpkt_timeout_), (int)nb_msgs);
}
nb_msgs = rtrd->nb_msgs();
// Update the stat for video fps.
// @remark https://github.com/ossrs/srs/issues/851
SrsStatistic *stat = _srs_stat;
if ((err = stat->on_video_frames(req, (int)(rtrd->nb_video_frames() - nb_frames))) != srs_success) {
return srs_error_wrap(err, "rtmp: stat video frames");
}
nb_frames = rtrd->nb_video_frames();
// reportable
if (pprint->can_print()) {
kbps_->sample();
bool mr = _srs_config->get_mr_enabled(req->vhost_);
srs_utime_t mr_sleep = _srs_config->get_mr_sleep(req->vhost_);
srs_trace("<- " SRS_CONSTS_LOG_CLIENT_PUBLISH " time=%d, okbps=%d,%d,%d, ikbps=%d,%d,%d, mr=%d/%d, p1stpt=%d, pnt=%d",
(int)pprint->age(), kbps_->get_send_kbps(), kbps_->get_send_kbps_30s(), kbps_->get_send_kbps_5m(),
kbps_->get_recv_kbps(), kbps_->get_recv_kbps_30s(), kbps_->get_recv_kbps_5m(), mr, srsu2msi(mr_sleep),
srsu2msi(publish_1stpkt_timeout_), srsu2msi(publish_normal_timeout_));
#ifdef SRS_APM
// TODO: Do not use pithy print for frame span.
ISrsApmSpan *sample = _srs_apm->span("publish-frame")->set_kind(SrsApmKindConsumer)->as_child(span.get())->attr("msgs", srs_fmt_sprintf("%" PRId64, nb_frames))->attr("kbps", srs_fmt_sprintf("%d", kbps_->get_recv_kbps_30s()));
srs_freep(sample);
#endif
}
}
return err;
}
srs_error_t SrsRtmpConn::acquire_publish(SrsSharedPtr<SrsLiveSource> source)
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
// Check whether RTMP stream is busy.
if (!source->can_publish(info_->edge_)) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp: stream %s is busy", req->get_stream_url().c_str());
}
// Check whether RTC stream is busy.
SrsSharedPtr<SrsRtcSource> rtc;
bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost_);
if (rtc_server_enabled && rtc_enabled && !info_->edge_) {
if ((err = _srs_rtc_sources->fetch_or_create(req, rtc)) != srs_success) {
return srs_error_wrap(err, "create source");
}
if (!rtc->can_publish()) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
}
}
// Check whether SRT stream is busy.
bool srt_server_enabled = _srs_config->get_srt_enabled();
bool srt_enabled = _srs_config->get_srt_enabled(req->vhost_);
if (srt_server_enabled && srt_enabled && !info_->edge_) {
SrsSharedPtr<SrsSrtSource> srt;
if ((err = _srs_srt_sources->fetch_or_create(req, srt)) != srs_success) {
return srs_error_wrap(err, "create source");
}
if (!srt->can_publish()) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "srt stream %s busy", req->get_stream_url().c_str());
}
}
#ifdef SRS_RTSP
// RTSP only support viewer, so we don't need to check it.
SrsSharedPtr<SrsRtspSource> rtsp;
bool rtsp_server_enabled = _srs_config->get_rtsp_server_enabled();
bool rtsp_enabled = _srs_config->get_rtsp_enabled(req->vhost_);
if (rtsp_server_enabled && rtsp_enabled && !info_->edge_) {
if ((err = _srs_rtsp_sources->fetch_or_create(req, rtsp)) != srs_success) {
return srs_error_wrap(err, "create source");
}
}
#endif
// Bridge to RTC streaming.
// TODO: FIXME: Need to convert RTMP to SRT.
SrsRtmpBridge *bridge = new SrsRtmpBridge();
#if defined(SRS_FFMPEG_FIT)
if (rtc.get() && _srs_config->get_rtc_from_rtmp(req->vhost_)) {
bridge->enable_rtmp2rtc(rtc);
}
#endif
#ifdef SRS_RTSP
if (rtsp.get() && _srs_config->get_rtsp_from_rtmp(req->vhost_)) {
bridge->enable_rtmp2rtsp(rtsp);
}
#endif
if (bridge->empty()) {
srs_freep(bridge);
} else if ((err = bridge->initialize(req)) != srs_success) {
srs_freep(bridge);
return srs_error_wrap(err, "bridge init");
}
source->set_bridge(bridge);
// Start publisher now.
if (info_->edge_) {
err = source->on_edge_start_publish();
} else {
err = source->on_publish();
}
return err;
}
void SrsRtmpConn::release_publish(SrsSharedPtr<SrsLiveSource> source)
{
// when edge, notice edge to change state.
// when origin, notice all service to unpublish.
if (info_->edge_) {
source->on_edge_proxy_unpublish();
} else {
source->on_unpublish();
}
}
srs_error_t SrsRtmpConn::handle_publish_message(SrsSharedPtr<SrsLiveSource> &source, SrsRtmpCommonMessage *msg)
{
srs_error_t err = srs_success;
// process publish event.
if (msg->header_.is_amf0_command() || msg->header_.is_amf3_command()) {
SrsRtmpCommand *pkt_raw = NULL;
if ((err = rtmp_->decode_message(msg, &pkt_raw)) != srs_success) {
return srs_error_wrap(err, "rtmp: decode message");
}
SrsUniquePtr<SrsRtmpCommand> pkt(pkt_raw);
// for flash, any packet is republish.
if (info_->type_ == SrsRtmpConnFlashPublish) {
// flash unpublish.
// TODO: maybe need to support republish.
srs_trace("flash flash publish finished.");
return srs_error_new(ERROR_CONTROL_REPUBLISH, "rtmp: republish");
}
// for fmle, drop others except the fmle start packet.
if (dynamic_cast<SrsFMLEStartPacket *>(pkt.get())) {
SrsFMLEStartPacket *unpublish = dynamic_cast<SrsFMLEStartPacket *>(pkt.get());
if ((err = rtmp_->fmle_unpublish(info_->res_->stream_id_, unpublish->transaction_id_)) != srs_success) {
return srs_error_wrap(err, "rtmp: republish");
}
return srs_error_new(ERROR_CONTROL_REPUBLISH, "rtmp: republish");
}
srs_trace("fmle ignore AMF0/AMF3 command message.");
return err;
}
// video, audio, data message
if ((err = process_publish_message(source, msg)) != srs_success) {
return srs_error_wrap(err, "rtmp: consume message");
}
return err;
}
srs_error_t SrsRtmpConn::process_publish_message(SrsSharedPtr<SrsLiveSource> &source, SrsRtmpCommonMessage *msg)
{
srs_error_t err = srs_success;
// for edge, directly proxy message to origin.
if (info_->edge_) {
if ((err = source->on_edge_proxy_publish(msg)) != srs_success) {
return srs_error_wrap(err, "rtmp: proxy publish");
}
return err;
}
// process audio packet
if (msg->header_.is_audio()) {
if ((err = source->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "rtmp: consume audio");
}
return err;
}
// process video packet
if (msg->header_.is_video()) {
if ((err = source->on_video(msg)) != srs_success) {
return srs_error_wrap(err, "rtmp: consume video");
}
return err;
}
// process aggregate packet
if (msg->header_.is_aggregate()) {
if ((err = source->on_aggregate(msg)) != srs_success) {
return srs_error_wrap(err, "rtmp: consume aggregate");
}
return err;
}
// process onMetaData
if (msg->header_.is_amf0_data() || msg->header_.is_amf3_data()) {
SrsRtmpCommand *pkt_raw = NULL;
if ((err = rtmp_->decode_message(msg, &pkt_raw)) != srs_success) {
return srs_error_wrap(err, "rtmp: decode message");
}
SrsUniquePtr<SrsRtmpCommand> pkt(pkt_raw);
if (dynamic_cast<SrsOnMetaDataPacket *>(pkt.get())) {
SrsOnMetaDataPacket *metadata = dynamic_cast<SrsOnMetaDataPacket *>(pkt.get());
if ((err = source->on_meta_data(msg, metadata)) != srs_success) {
return srs_error_wrap(err, "rtmp: consume metadata");
}
return err;
}
return err;
}
return err;
}
srs_error_t SrsRtmpConn::process_play_control_msg(SrsLiveConsumer *consumer, SrsRtmpCommonMessage *msg_raw)
{
srs_error_t err = srs_success;
if (!msg_raw) {
return err;
}
SrsUniquePtr<SrsRtmpCommonMessage> msg(msg_raw);
if (!msg->header_.is_amf0_command() && !msg->header_.is_amf3_command()) {
return err;
}
SrsRtmpCommand *pkt_raw = NULL;
if ((err = rtmp_->decode_message(msg.get(), &pkt_raw)) != srs_success) {
return srs_error_wrap(err, "rtmp: decode message");
}
SrsUniquePtr<SrsRtmpCommand> pkt(pkt_raw);
// for jwplayer/flowplayer, which send close as pause message.
SrsCloseStreamPacket *close = dynamic_cast<SrsCloseStreamPacket *>(pkt.get());
if (close) {
return srs_error_new(ERROR_CONTROL_RTMP_CLOSE, "rtmp: close stream");
}
// call msg,
// support response null first,
// TODO: FIXME: response in right way, or forward in edge mode.
SrsCallPacket *call = dynamic_cast<SrsCallPacket *>(pkt.get());
if (call) {
// only response it when transaction id not zero,
// for the zero means donot need response.
if (call->transaction_id_ > 0) {
SrsCallResPacket *res = new SrsCallResPacket(call->transaction_id_);
res->command_object_ = SrsAmf0Any::null();
res->response_ = SrsAmf0Any::null();
if ((err = rtmp_->send_and_free_packet(res, 0)) != srs_success) {
return srs_error_wrap(err, "rtmp: send packets");
}
}
return err;
}
// pause
SrsPausePacket *pause = dynamic_cast<SrsPausePacket *>(pkt.get());
if (pause) {
if ((err = rtmp_->on_play_client_pause(info_->res_->stream_id_, pause->is_pause_)) != srs_success) {
return srs_error_wrap(err, "rtmp: pause");
}
if ((err = consumer->on_play_client_pause(pause->is_pause_)) != srs_success) {
return srs_error_wrap(err, "rtmp: pause");
}
return err;
}
// other msg.
return err;
}
void SrsRtmpConn::set_sock_options()
{
ISrsRequest *req = info_->req_;
bool nvalue = _srs_config->get_tcp_nodelay(req->vhost_);
if (nvalue != tcp_nodelay_) {
tcp_nodelay_ = nvalue;
srs_error_t err = transport_->set_tcp_nodelay(tcp_nodelay_);
if (err != srs_success) {
srs_warn("ignore err %s", srs_error_desc(err).c_str());
srs_freep(err);
}
}
}
srs_error_t SrsRtmpConn::check_edge_token_traverse_auth()
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
srs_assert(req);
vector<string> args = _srs_config->get_vhost_edge_origin(req->vhost_)->args_;
if (args.empty()) {
return err;
}
for (int i = 0; i < (int)args.size(); i++) {
string hostport = args.at(i);
// select the origin.
string server;
int port = SRS_CONSTS_RTMP_DEFAULT_PORT;
srs_net_split_hostport(hostport, server, port);
SrsUniquePtr<SrsTcpClient> transport(new SrsTcpClient(server, port, SRS_EDGE_TOKEN_TRAVERSE_TIMEOUT));
if ((err = transport->connect()) != srs_success) {
srs_warn("Illegal edge token, tcUrl=%s, %s", req->tcUrl_.c_str(), srs_error_desc(err).c_str());
srs_freep(err);
continue;
}
SrsUniquePtr<SrsRtmpClient> client(new SrsRtmpClient(transport.get()));
return do_token_traverse_auth(client.get());
}
return srs_error_new(ERROR_EDGE_PORT_INVALID, "rtmp: Illegal edge token, server=%d", (int)args.size());
}
srs_error_t SrsRtmpConn::do_token_traverse_auth(SrsRtmpClient *client)
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
srs_assert(client);
client->set_recv_timeout(SRS_CONSTS_RTMP_TIMEOUT);
client->set_send_timeout(SRS_CONSTS_RTMP_TIMEOUT);
if ((err = client->handshake()) != srs_success) {
return srs_error_wrap(err, "rtmp: handshake");
}
// for token tranverse, always take the debug info(which carries token).
SrsServerInfo si;
if ((err = client->connect_app(req->app_, req->tcUrl_, req, true, &si)) != srs_success) {
return srs_error_wrap(err, "rtmp: connect tcUrl");
}
srs_trace("edge token auth ok, tcUrl=%s", req->tcUrl_.c_str());
return err;
}
srs_error_t SrsRtmpConn::on_disconnect()
{
srs_error_t err = srs_success;
http_hooks_on_close();
// TODO: FIXME: Implements it.
return err;
}
srs_error_t SrsRtmpConn::http_hooks_on_connect()
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost_)) {
return err;
}
// the http hooks will cause context switch,
// so we must copy all hooks for the on_connect may freed.
// @see https://github.com/ossrs/srs/issues/475
vector<string> hooks;
if (true) {
SrsConfDirective *conf = _srs_config->get_vhost_on_connect(req->vhost_);
if (!conf) {
return err;
}
hooks = conf->args_;
}
for (int i = 0; i < (int)hooks.size(); i++) {
std::string url = hooks.at(i);
if ((err = _srs_hooks->on_connect(url, req)) != srs_success) {
return srs_error_wrap(err, "rtmp on_connect %s", url.c_str());
}
}
return err;
}
void SrsRtmpConn::http_hooks_on_close()
{
ISrsRequest *req = info_->req_;
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost_)) {
return;
}
// the http hooks will cause context switch,
// so we must copy all hooks for the on_connect may freed.
// @see https://github.com/ossrs/srs/issues/475
vector<string> hooks;
if (true) {
SrsConfDirective *conf = _srs_config->get_vhost_on_close(req->vhost_);
if (!conf) {
return;
}
hooks = conf->args_;
}
for (int i = 0; i < (int)hooks.size(); i++) {
std::string url = hooks.at(i);
_srs_hooks->on_close(url, req, transport_->get_send_bytes(), transport_->get_recv_bytes());
}
}
srs_error_t SrsRtmpConn::http_hooks_on_publish()
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost_)) {
return err;
}
// the http hooks will cause context switch,
// so we must copy all hooks for the on_connect may freed.
// @see https://github.com/ossrs/srs/issues/475
vector<string> hooks;
if (true) {
SrsConfDirective *conf = _srs_config->get_vhost_on_publish(req->vhost_);
if (!conf) {
return err;
}
hooks = conf->args_;
}
for (int i = 0; i < (int)hooks.size(); i++) {
std::string url = hooks.at(i);
if ((err = _srs_hooks->on_publish(url, req)) != srs_success) {
return srs_error_wrap(err, "rtmp on_publish %s", url.c_str());
}
}
return err;
}
void SrsRtmpConn::http_hooks_on_unpublish()
{
ISrsRequest *req = info_->req_;
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost_)) {
return;
}
// the http hooks will cause context switch,
// so we must copy all hooks for the on_connect may freed.
// @see https://github.com/ossrs/srs/issues/475
vector<string> hooks;
if (true) {
SrsConfDirective *conf = _srs_config->get_vhost_on_unpublish(req->vhost_);
if (!conf) {
return;
}
hooks = conf->args_;
}
for (int i = 0; i < (int)hooks.size(); i++) {
std::string url = hooks.at(i);
_srs_hooks->on_unpublish(url, req);
}
}
srs_error_t SrsRtmpConn::http_hooks_on_play()
{
srs_error_t err = srs_success;
ISrsRequest *req = info_->req_;
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost_)) {
return err;
}
// the http hooks will cause context switch,
// so we must copy all hooks for the on_connect may freed.
// @see https://github.com/ossrs/srs/issues/475
vector<string> hooks;
if (true) {
SrsConfDirective *conf = _srs_config->get_vhost_on_play(req->vhost_);
if (!conf) {
return err;
}
hooks = conf->args_;
}
for (int i = 0; i < (int)hooks.size(); i++) {
std::string url = hooks.at(i);
if ((err = _srs_hooks->on_play(url, req)) != srs_success) {
return srs_error_wrap(err, "rtmp on_play %s", url.c_str());
}
}
return err;
}
void SrsRtmpConn::http_hooks_on_stop()
{
ISrsRequest *req = info_->req_;
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost_)) {
return;
}
// the http hooks will cause context switch,
// so we must copy all hooks for the on_connect may freed.
// @see https://github.com/ossrs/srs/issues/475
vector<string> hooks;
if (true) {
SrsConfDirective *conf = _srs_config->get_vhost_on_stop(req->vhost_);
if (!conf) {
return;
}
hooks = conf->args_;
}
for (int i = 0; i < (int)hooks.size(); i++) {
std::string url = hooks.at(i);
_srs_hooks->on_stop(url, req);
}
return;
}
srs_error_t SrsRtmpConn::start()
{
srs_error_t err = srs_success;
if ((err = trd_->start()) != srs_success) {
return srs_error_wrap(err, "coroutine");
}
return err;
}
srs_error_t SrsRtmpConn::cycle()
{
srs_error_t err = srs_success;
// Serve the client.
err = do_cycle();
// Update statistic when done.
SrsStatistic *stat = _srs_stat;
stat->kbps_add_delta(get_id().c_str(), delta_);
stat->on_disconnect(get_id().c_str(), err);
// Notify manager to remove it.
// Note that we create this object, so we use manager to remove it.
manager_->remove(this);
// success.
if (err == srs_success) {
srs_trace("client finished.");
return err;
}
// It maybe success with message.
if (srs_error_code(err) == ERROR_SUCCESS) {
srs_trace("client finished%s.", srs_error_summary(err).c_str());
srs_freep(err);
return err;
}
// client close peer.
// TODO: FIXME: Only reset the error when client closed it.
if (srs_is_client_gracefully_close(err)) {
srs_warn("client disconnect peer. ret=%d", srs_error_code(err));
} else if (srs_is_server_gracefully_close(err)) {
srs_warn("server disconnect. ret=%d", srs_error_code(err));
} else {
srs_error("serve error %s", srs_error_desc(err).c_str());
}
srs_freep(err);
return srs_success;
}
string SrsRtmpConn::remote_ip()
{
return ip_;
}
const SrsContextId &SrsRtmpConn::get_id()
{
return trd_->cid();
}
void SrsRtmpConn::expire()
{
trd_->interrupt();
}