mirror of
https://github.com/ossrs/srs.git
synced 2025-02-15 04:42:04 +00:00
1407 lines
46 KiB
C++
1407 lines
46 KiB
C++
/**
|
|
* The MIT License (MIT)
|
|
*
|
|
* Copyright (c) 2013-2020 Winlin
|
|
*
|
|
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
|
* this software and associated documentation files (the "Software"), to deal in
|
|
* the Software without restriction, including without limitation the rights to
|
|
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
|
|
* the Software, and to permit persons to whom the Software is furnished to do so,
|
|
* subject to the following conditions:
|
|
*
|
|
* The above copyright notice and this permission notice shall be included in all
|
|
* copies or substantial portions of the Software.
|
|
*
|
|
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
|
|
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
|
|
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
|
|
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
|
|
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
|
*/
|
|
|
|
#include <srs_app_rtmp_conn.hpp>
|
|
|
|
#include <stdlib.h>
|
|
#include <sys/socket.h>
|
|
#include <netinet/in.h>
|
|
#include <arpa/inet.h>
|
|
#include <netinet/tcp.h>
|
|
#include <unistd.h>
|
|
using namespace std;
|
|
|
|
#include <srs_kernel_error.hpp>
|
|
#include <srs_kernel_log.hpp>
|
|
#include <srs_rtmp_stack.hpp>
|
|
#include <srs_core_autofree.hpp>
|
|
#include <srs_app_source.hpp>
|
|
#include <srs_app_server.hpp>
|
|
#include <srs_app_pithy_print.hpp>
|
|
#include <srs_app_config.hpp>
|
|
#include <srs_app_refer.hpp>
|
|
#include <srs_app_hls.hpp>
|
|
#include <srs_app_bandwidth.hpp>
|
|
#include <srs_app_st.hpp>
|
|
#include <srs_app_http_hooks.hpp>
|
|
#include <srs_app_edge.hpp>
|
|
#include <srs_app_utility.hpp>
|
|
#include <srs_rtmp_msg_array.hpp>
|
|
#include <srs_protocol_amf0.hpp>
|
|
#include <srs_app_recv_thread.hpp>
|
|
#include <srs_core_performance.hpp>
|
|
#include <srs_kernel_utility.hpp>
|
|
#include <srs_app_security.hpp>
|
|
#include <srs_app_statistic.hpp>
|
|
#include <srs_protocol_utility.hpp>
|
|
#include <srs_protocol_json.hpp>
|
|
|
|
// 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();
|
|
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);
|
|
}
|
|
|
|
SrsRtmpConn::SrsRtmpConn(SrsServer* svr, srs_netfd_t c, string cip, int port) : SrsConnection(svr, c, cip, port)
|
|
{
|
|
server = svr;
|
|
|
|
rtmp = new SrsRtmpServer(skt);
|
|
refer = new SrsRefer();
|
|
bandwidth = new SrsBandwidth();
|
|
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);
|
|
|
|
srs_freep(info);
|
|
srs_freep(rtmp);
|
|
srs_freep(refer);
|
|
srs_freep(bandwidth);
|
|
srs_freep(security);
|
|
}
|
|
|
|
void SrsRtmpConn::dispose()
|
|
{
|
|
SrsConnection::dispose();
|
|
|
|
// wakeup the handler which need to notice.
|
|
if (wakable) {
|
|
wakable->wakeup();
|
|
}
|
|
}
|
|
|
|
// TODO: return detail message when error for client.
|
|
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) {
|
|
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) {
|
|
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_error_reset(err);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::on_reload_vhost_removed(string vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
|
|
if (req->vhost != vhost) {
|
|
return err;
|
|
}
|
|
|
|
// if the vhost connected is removed, disconnect the client.
|
|
srs_trace("vhost %s removed/disabled, close client url=%s",
|
|
vhost.c_str(), req->get_stream_url().c_str());
|
|
|
|
// should never close the fd in another thread,
|
|
// one fd should managed by one thread, we should use interrupt instead.
|
|
// so we just ignore the vhost enabled event.
|
|
//srs_close_stfd(stfd);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::on_reload_vhost_play(string vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
|
|
if (req->vhost != vhost) {
|
|
return err;
|
|
}
|
|
|
|
// send_min_interval
|
|
if (true) {
|
|
srs_utime_t v = _srs_config->get_send_min_interval(vhost);
|
|
if (v != send_min_interval) {
|
|
srs_trace("apply smi %d=>%d ms", srsu2msi(send_min_interval), srsu2msi(v));
|
|
send_min_interval = v;
|
|
}
|
|
}
|
|
|
|
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime);
|
|
mw_sleep = _srs_config->get_mw_sleep(req->vhost);
|
|
set_socket_buffer(mw_sleep);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::on_reload_vhost_tcp_nodelay(string vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
|
|
if (req->vhost != vhost) {
|
|
return err;
|
|
}
|
|
|
|
set_sock_options();
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::on_reload_vhost_realtime(string vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
|
|
if (req->vhost != vhost) {
|
|
return err;
|
|
}
|
|
|
|
bool realtime_enabled = _srs_config->get_realtime_enabled(req->vhost);
|
|
if (realtime_enabled != realtime) {
|
|
srs_trace("realtime changed %d=>%d", realtime, realtime_enabled);
|
|
realtime = realtime_enabled;
|
|
}
|
|
|
|
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime);
|
|
mw_sleep = _srs_config->get_mw_sleep(req->vhost);
|
|
set_socket_buffer(mw_sleep);
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::on_reload_vhost_publish(string vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
|
|
if (req->vhost != vhost) {
|
|
return err;
|
|
}
|
|
|
|
srs_utime_t p1stpt = _srs_config->get_publish_1stpkt_timeout(req->vhost);
|
|
if (p1stpt != publish_1stpkt_timeout) {
|
|
srs_trace("p1stpt changed %d=>%d", srsu2msi(publish_1stpkt_timeout), srsu2msi(p1stpt));
|
|
publish_1stpkt_timeout = p1stpt;
|
|
}
|
|
|
|
srs_utime_t pnt = _srs_config->get_publish_normal_timeout(req->vhost);
|
|
if (pnt != publish_normal_timeout) {
|
|
srs_trace("pnt changed %d=>%d", srsu2msi(publish_normal_timeout), srsu2msi(pnt));
|
|
publish_normal_timeout = pnt;
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtmpConn::remark(int64_t* in, int64_t* out)
|
|
{
|
|
kbps->remark(in, out);
|
|
}
|
|
|
|
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.
|
|
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;
|
|
}
|
|
|
|
// 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.
|
|
// @see: https://github.com/ossrs/srs/issues/39
|
|
// 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;
|
|
|
|
SrsRequest* 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_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 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);
|
|
|
|
// find a source to serve.
|
|
SrsSource* source = NULL;
|
|
if ((err = _srs_sources->fetch_or_create(req, server, &source)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: fetch source");
|
|
}
|
|
srs_assert(source != NULL);
|
|
|
|
// update the statistic when source disconveried.
|
|
SrsStatistic* stat = SrsStatistic::instance();
|
|
if ((err = stat->on_client(_srs_context->get_id(), req, this, info->type)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: stat client");
|
|
}
|
|
|
|
bool enabled_cache = _srs_config->get_gop_cache(req->vhost);
|
|
srs_trace("source url=%s, ip=%s, cache=%d, is_edge=%d, source_id=[%d][%s]",
|
|
req->get_stream_url().c_str(), ip.c_str(), enabled_cache, info->edge, ::getpid(), source->source_id().c_str());
|
|
source->set_cache(enabled_cache);
|
|
|
|
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");
|
|
}
|
|
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;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::check_vhost(bool try_default_vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* 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(SrsSource* source)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// Check page referer of player.
|
|
SrsRequest* 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 = SrsHttpHooks::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_generate_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_error_reset(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.
|
|
SrsConsumer* consumer = NULL;
|
|
SrsAutoFree(SrsConsumer, consumer);
|
|
if ((err = source->create_consumer(this, consumer)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: create consumer");
|
|
}
|
|
if ((err = source->consumer_dumps(consumer)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: dumps consumer");
|
|
}
|
|
|
|
// Use receiving thread to receive packets from peer.
|
|
// @see: https://github.com/ossrs/srs/issues/217
|
|
SrsQueueRecvThread trd(consumer, 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;
|
|
err = do_playing(source, consumer, &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(SrsSource* source, SrsConsumer* consumer, SrsQueueRecvThread* rtrd)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
srs_assert(req);
|
|
srs_assert(consumer);
|
|
|
|
// initialize other components
|
|
SrsPithyPrint* pprint = SrsPithyPrint::create_rtmp_play();
|
|
SrsAutoFree(SrsPithyPrint, pprint);
|
|
|
|
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);
|
|
// setup the mw config.
|
|
// when mw_sleep changed, resize the socket send buffer.
|
|
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime);
|
|
mw_sleep = _srs_config->get_mw_sleep(req->vhost);
|
|
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.
|
|
// @see: https://github.com/ossrs/srs/issues/196
|
|
// @see: https://github.com/ossrs/srs/issues/217
|
|
while (!rtrd->empty()) {
|
|
SrsCommonMessage* 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/251
|
|
// @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);
|
|
}
|
|
|
|
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++) {
|
|
SrsSharedPtrMessage* 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);
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::publishing(SrsSource* 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/237
|
|
SrsPublishRecvThread 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(SrsSource* source, SrsPublishRecvThread* rtrd)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
SrsPithyPrint* pprint = SrsPithyPrint::create_rtmp_publish();
|
|
SrsAutoFree(SrsPithyPrint, pprint);
|
|
|
|
// start isolate recv thread.
|
|
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);
|
|
|
|
// 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);
|
|
}
|
|
|
|
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");
|
|
}
|
|
|
|
pprint->elapse();
|
|
|
|
// cond wait for timeout.
|
|
if (nb_msgs == 0) {
|
|
// when not got msgs, wait for a larger timeout.
|
|
// @see https://github.com/ossrs/srs/issues/441
|
|
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 = SrsStatistic::instance();
|
|
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));
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::acquire_publish(SrsSource* source)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* req = info->req;
|
|
|
|
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());
|
|
}
|
|
|
|
// when edge, ignore the publish event, directly proxy it.
|
|
if (info->edge) {
|
|
if ((err = source->on_edge_start_publish()) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: edge start publish");
|
|
}
|
|
} else {
|
|
if ((err = source->on_publish()) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: source publish");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsRtmpConn::release_publish(SrsSource* 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(SrsSource* source, SrsCommonMessage* msg)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// process publish event.
|
|
if (msg->header.is_amf0_command() || msg->header.is_amf3_command()) {
|
|
SrsPacket* pkt = NULL;
|
|
if ((err = rtmp->decode_message(msg, &pkt)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: decode message");
|
|
}
|
|
SrsAutoFree(SrsPacket, pkt);
|
|
|
|
// 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)) {
|
|
SrsFMLEStartPacket* unpublish = dynamic_cast<SrsFMLEStartPacket*>(pkt);
|
|
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(SrsSource* source, SrsCommonMessage* 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()) {
|
|
SrsPacket* pkt = NULL;
|
|
if ((err = rtmp->decode_message(msg, &pkt)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: decode message");
|
|
}
|
|
SrsAutoFree(SrsPacket, pkt);
|
|
|
|
if (dynamic_cast<SrsOnMetaDataPacket*>(pkt)) {
|
|
SrsOnMetaDataPacket* metadata = dynamic_cast<SrsOnMetaDataPacket*>(pkt);
|
|
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(SrsConsumer* consumer, SrsCommonMessage* msg)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (!msg) {
|
|
return err;
|
|
}
|
|
SrsAutoFree(SrsCommonMessage, msg);
|
|
|
|
if (!msg->header.is_amf0_command() && !msg->header.is_amf3_command()) {
|
|
return err;
|
|
}
|
|
|
|
SrsPacket* pkt = NULL;
|
|
if ((err = rtmp->decode_message(msg, &pkt)) != srs_success) {
|
|
return srs_error_wrap(err, "rtmp: decode message");
|
|
}
|
|
SrsAutoFree(SrsPacket, pkt);
|
|
|
|
// for jwplayer/flowplayer, which send close as pause message.
|
|
// @see https://github.com/ossrs/srs/issues/6
|
|
SrsCloseStreamPacket* close = dynamic_cast<SrsCloseStreamPacket*>(pkt);
|
|
if (close) {
|
|
return srs_error_new(ERROR_CONTROL_RTMP_CLOSE, "rtmp: close stream");
|
|
}
|
|
|
|
// 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) {
|
|
// 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);
|
|
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()
|
|
{
|
|
SrsRequest* req = info->req;
|
|
|
|
bool nvalue = _srs_config->get_tcp_nodelay(req->vhost);
|
|
if (nvalue != tcp_nodelay) {
|
|
tcp_nodelay = nvalue;
|
|
|
|
srs_error_t err = 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;
|
|
|
|
SrsRequest* 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_parse_hostport(hostport, server, port);
|
|
|
|
SrsTcpClient* transport = new SrsTcpClient(server, port, SRS_EDGE_TOKEN_TRAVERSE_TIMEOUT);
|
|
SrsAutoFree(SrsTcpClient, transport);
|
|
|
|
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;
|
|
}
|
|
|
|
SrsRtmpClient* client = new SrsRtmpClient(transport);
|
|
SrsAutoFree(SrsRtmpClient, client);
|
|
return do_token_traverse_auth(client);
|
|
}
|
|
|
|
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;
|
|
|
|
SrsRequest* 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;
|
|
|
|
SrsRequest* 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 = SrsHttpHooks::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()
|
|
{
|
|
SrsRequest* 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);
|
|
SrsHttpHooks::on_close(url, req, kbps->get_send_bytes(), kbps->get_recv_bytes());
|
|
}
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::http_hooks_on_publish()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* 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 = SrsHttpHooks::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()
|
|
{
|
|
SrsRequest* 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);
|
|
SrsHttpHooks::on_unpublish(url, req);
|
|
}
|
|
}
|
|
|
|
srs_error_t SrsRtmpConn::http_hooks_on_play()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRequest* 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 = SrsHttpHooks::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()
|
|
{
|
|
SrsRequest* 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);
|
|
SrsHttpHooks::on_stop(url, req);
|
|
}
|
|
|
|
return;
|
|
}
|
|
|