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For Edge, only support RTMP or HTTP-FLV. v7.0.94 (#4513)

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Winlin 2025-09-27 19:35:34 -04:00 committed by GitHub
parent c0fc8cb093
commit c7821b4770
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GPG Key ID: B5690EEEBB952194
9 changed files with 176 additions and 23 deletions
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66
trunk/conf/edge.conf
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@ -15,6 +15,26 @@ http_server {
listen 8080;
dir ./objs/nginx/html;
}
http_api {
enabled on;
listen 1985;
}
# Edge does not support WebRTC, so even if enabled in config,
# it will be automatically disabled at runtime.
rtc_server {
enabled on;
listen 8000;
candidate $CANDIDATE;
}
# Edge does not support SRT, so even if enabled in config,
# it will be automatically disabled at runtime.
srt_server {
enabled on;
listen 10080;
}
vhost __defaultVhost__ {
cluster {
mode remote;
@ -24,4 +44,50 @@ vhost __defaultVhost__ {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
# Edge does not support WebRTC, so even if enabled in config,
# it will be automatically disabled at runtime.
rtc {
enabled on;
rtmp_to_rtc on;
rtc_to_rtmp on;
}
# Edge does not support SRT, so even if enabled in config,
# it will be automatically disabled at runtime.
srt {
enabled on;
srt_to_rtmp on;
}
# Edge does not support HDS, so even if enabled in config,
# it will be automatically disabled at runtime.
hds {
enabled on;
}
# Edge does not support HLS, so even if enabled in config,
# it will be automatically disabled at runtime.
hls {
enabled on;
}
# Edge does not support MPEG-DASH, so even if enabled in config,
# it will be automatically disabled at runtime.
dash {
enabled on;
}
# Edge does not support forwarding, so even if enabled in config,
# it will be automatically disabled at runtime.
forward {
enabled on;
destination 127.0.0.1:19350;
}
# Edge does not support DVR, so even if enabled in config,
# it will be automatically disabled at runtime.
dvr {
enabled on;
}
}

4
trunk/doc/CHANGELOG.md
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@ -7,6 +7,7 @@ The changelog for SRS.
<a name="v7-changes"></a>
## SRS 7.0 Changelog
* v7.0, 2025-09-27, Merge [#4513](https://github.com/ossrs/srs/pull/4513): For Edge, only support RTMP or HTTP-FLV. v7.0.94 (#4513)
* v7.0, 2025-09-21, Merge [#4505](https://github.com/ossrs/srs/pull/4505): improve blackbox test for rtsp. v7.0.93 (#4505)
* v7.0, 2025-09-21, Fix WHIP with transcoding bug. v7.0.92 (#4495)
* v7.0, 2025-09-20, Merge [#4504](https://github.com/ossrs/srs/pull/4504): fix rtsp compiling warning. v7.0.91 (#4504)
@ -107,7 +108,8 @@ The changelog for SRS.
<a name="v6-changes"></a>
## SRS 6.0 Changelog
* v6.0, 2025-09-21, Fix WHIP with transcoding bug. v6.0.179 (#4495)
* v6.0, 2025-09-27, For Edge, only support RTMP or HTTP-FLV. v6.0.179 (#4512)
* v6.0, 2025-09-21, Fix WHIP with transcoding bug. v6.0.178 (#4495)
* v6.0, 2025-09-15, RTC2RTMP: Fix sequence number wraparound assertion crashes. v6.0.177 (#4491)
* v6.0, 2025-09-05, RTX: Fix race condition for timer. v6.0.176 (#4470) (#4474)
* v6.0, 2025-08-26, Merge [#4451](https://github.com/ossrs/srs/pull/4451): RTC: Fix null pointer crash in RTC2RTMP when start packet is missing. v6.0.175 (#4451)

22
trunk/src/app/srs_app_rtc_api.cpp
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@ -206,6 +206,13 @@ srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter *w, ISrsHttpMessa
// Whether enabled.
bool server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(ruc->req_->vhost_);
bool edge = _srs_config->get_vhost_is_edge(ruc->req_->vhost_);
if (rtc_enabled && edge) {
rtc_enabled = false;
srs_warn("disable WebRTC for edge vhost=%s", ruc->req_->vhost_.c_str());
}
if (server_enabled && !rtc_enabled) {
srs_warn("RTC disabled in vhost %s", ruc->req_->vhost_.c_str());
}
@ -222,7 +229,13 @@ srs_error_t SrsGoApiRtcPlay::serve_http(ISrsHttpResponseWriter *w, ISrsHttpMessa
}
// For RTMP to RTC, fail if disabled and RTMP is active, see https://github.com/ossrs/srs/issues/2728
if (!is_rtc_stream_active && !_srs_config->get_rtc_from_rtmp(ruc->req_->vhost_)) {
bool rtmp_to_rtc = _srs_config->get_rtc_from_rtmp(ruc->req_->vhost_);
if (rtmp_to_rtc && edge) {
rtmp_to_rtc = false;
srs_warn("disable RTMP to WebRTC for edge vhost=%s", ruc->req_->vhost_.c_str());
}
if (!is_rtc_stream_active && !rtmp_to_rtc) {
SrsSharedPtr<SrsLiveSource> live_source = _srs_sources->fetch(ruc->req_);
if (live_source.get() && !live_source->inactive()) {
return srs_error_new(ERROR_RTC_DISABLED, "Disabled rtmp_to_rtc of %s, see #2728", ruc->req_->vhost_.c_str());
@ -494,6 +507,13 @@ srs_error_t SrsGoApiRtcPublish::serve_http(ISrsHttpResponseWriter *w, ISrsHttpMe
// Whether enabled.
bool server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(ruc->req_->vhost_);
bool edge = _srs_config->get_vhost_is_edge(ruc->req_->vhost_);
if (rtc_enabled && edge) {
rtc_enabled = false;
srs_warn("disable WebRTC for edge vhost=%s", ruc->req_->vhost_.c_str());
}
if (server_enabled && !rtc_enabled) {
srs_warn("RTC disabled in vhost %s", ruc->req_->vhost_.c_str());
}

7
trunk/src/app/srs_app_rtc_conn.cpp
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@ -1280,6 +1280,13 @@ srs_error_t SrsRtcPublishStream::initialize(ISrsRequest *r, SrsRtcSourceDescript
// Bridge to RTMP.
// TODO: Support bridge to RTSP.
bool rtc_to_rtmp = config_->get_rtc_to_rtmp(req_->vhost_);
bool edge = _srs_config->get_vhost_is_edge(req_->vhost_);
if (rtc_to_rtmp && edge) {
rtc_to_rtmp = false;
srs_warn("disable WebRTC to RTMP for edge vhost=%s", req_->vhost_.c_str());
}
if (rtc_to_rtmp) {
// Disable GOP cache for RTC2RTMP bridge, to keep the streams in sync,
// especially for stream merging.

15
trunk/src/app/srs_app_rtmp_conn.cpp
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@ -1021,6 +1021,13 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsSharedPtr<SrsLiveSource> source)
SrsSharedPtr<SrsRtcSource> rtc;
bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost_);
bool edge = _srs_config->get_vhost_is_edge(req->vhost_);
if (rtc_enabled && edge) {
rtc_enabled = false;
srs_warn("disable WebRTC for edge vhost=%s", req->vhost_.c_str());
}
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");
@ -1062,7 +1069,13 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsSharedPtr<SrsLiveSource> source)
SrsRtmpBridge *bridge = new SrsRtmpBridge();
#if defined(SRS_FFMPEG_FIT)
if (rtc.get() && _srs_config->get_rtc_from_rtmp(req->vhost_)) {
bool rtmp_to_rtc = _srs_config->get_rtc_from_rtmp(req->vhost_);
if (rtmp_to_rtc && edge) {
rtmp_to_rtc = false;
srs_warn("disable RTMP to WebRTC for edge vhost=%s", req->vhost_.c_str());
}
if (rtc.get() && rtmp_to_rtc) {
bridge->enable_rtmp2rtc(rtc);
}
#endif

47
trunk/src/app/srs_app_rtmp_source.cpp
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@ -1719,7 +1719,7 @@ SrsLiveSource::SrsLiveSource()
play_edge_ = new SrsPlayEdge();
publish_edge_ = new SrsPublishEdge();
gop_cache_ = new SrsGopCache();
hub_ = new SrsOriginHub();
hub_ = NULL;
meta_ = new SrsMetaCache();
format_ = new SrsRtmpFormat();
@ -1758,15 +1758,18 @@ SrsLiveSource::~SrsLiveSource()
void SrsLiveSource::dispose()
{
hub_->dispose();
if (hub_) {
hub_->dispose();
}
meta_->dispose();
gop_cache_->dispose();
}
srs_error_t SrsLiveSource::cycle()
{
srs_error_t err = hub_->cycle();
if (err != srs_success) {
srs_error_t err = srs_success;
if (hub_ && (err = hub_->cycle()) != srs_success) {
return srs_error_wrap(err, "hub cycle");
}
@ -1792,7 +1795,7 @@ bool SrsLiveSource::stream_is_dead()
}
// Origin hub delay cleanup.
if (now < stream_die_at_ + hub_->cleanup_delay()) {
if (hub_ && now < stream_die_at_ + hub_->cleanup_delay()) {
return false;
}
@ -1853,7 +1856,16 @@ srs_error_t SrsLiveSource::initialize(SrsSharedPtr<SrsLiveSource> wrapper, ISrsR
srs_utime_t queue_size = _srs_config->get_queue_length(req_->vhost_);
publish_edge_->set_queue_size(queue_size);
if ((err = hub_->initialize(wrapper, req_)) != srs_success) {
// Create and initialize origin hub only for origin servers, not edge servers
bool edge = _srs_config->get_vhost_is_edge(req_->vhost_);
if (!edge) {
srs_freep(hub_);
hub_ = new SrsOriginHub();
} else {
srs_warn("disable OriginHub creation for edge vhost=%s", req_->vhost_.c_str());
}
if (hub_ && (err = hub_->initialize(wrapper, req_)) != srs_success) {
return srs_error_wrap(err, "hub");
}
@ -1967,7 +1979,11 @@ srs_error_t SrsLiveSource::on_meta_data(SrsRtmpCommonMessage *msg, SrsOnMetaData
}
// Copy to hub to all utilities.
return hub_->on_meta_data(meta_->data(), metadata);
if (hub_ && (err = hub_->on_meta_data(meta_->data(), metadata)) != srs_success) {
return srs_error_wrap(err, "hub consume metadata");
}
return err;
}
srs_error_t SrsLiveSource::on_audio(SrsRtmpCommonMessage *shared_audio)
@ -2052,7 +2068,7 @@ srs_error_t SrsLiveSource::on_audio_imp(SrsMediaPacket *msg)
}
// Copy to hub to all utilities.
if ((err = hub_->on_audio(msg)) != srs_success) {
if (hub_ && (err = hub_->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "consume audio");
}
@ -2173,7 +2189,7 @@ srs_error_t SrsLiveSource::on_video_imp(SrsMediaPacket *msg)
}
// Copy to hub to all utilities.
if ((err = hub_->on_video(msg, is_sequence_header)) != srs_success) {
if (hub_ && (err = hub_->on_video(msg, is_sequence_header)) != srs_success) {
return srs_error_wrap(err, "hub consume video");
}
@ -2329,7 +2345,7 @@ srs_error_t SrsLiveSource::on_publish()
last_packet_time_ = 0;
// Notify the hub about the publish event.
if ((err = hub_->on_publish()) != srs_success) {
if (hub_ && (err = hub_->on_publish()) != srs_success) {
return srs_error_wrap(err, "hub publish");
}
@ -2363,7 +2379,9 @@ void SrsLiveSource::on_unpublish()
}
// Notify the hub about the unpublish event.
hub_->on_unpublish();
if (hub_) {
hub_->on_unpublish();
}
// only clear the gop cache,
// donot clear the sequence header, for it maybe not changed,
@ -2450,7 +2468,8 @@ srs_error_t SrsLiveSource::consumer_dumps(SrsLiveConsumer *consumer, bool ds, bo
}
// If stream is publishing, dumps the sequence header and gop cache.
if (hub_->active()) {
bool hub_active = hub_ ? hub_->active() : false;
if (hub_active) {
// Copy metadata and sequence header to consumer.
if ((err = meta_->dumps(consumer, atc_, jitter_algorithm_, dm, ds)) != srs_success) {
return srs_error_wrap(err, "meta dumps");
@ -2464,9 +2483,9 @@ srs_error_t SrsLiveSource::consumer_dumps(SrsLiveConsumer *consumer, bool ds, bo
// print status.
if (dg) {
srs_trace("create consumer, active=%d, queue_size=%dms, jitter=%d", hub_->active(), srsu2msi(queue_size), jitter_algorithm_);
srs_trace("create consumer, active=%d, queue_size=%dms, jitter=%d", hub_active, srsu2msi(queue_size), jitter_algorithm_);
} else {
srs_trace("create consumer, active=%d, ignore gop cache, jitter=%d", hub_->active(), jitter_algorithm_);
srs_trace("create consumer, active=%d, ignore gop cache, jitter=%d", hub_active, jitter_algorithm_);
}
return err;

34
trunk/src/app/srs_app_srt_conn.cpp
View File

@ -278,7 +278,15 @@ srs_error_t SrsMpegtsSrtConn::do_cycle()
req_->vhost_ = parsed_vhost->arg0();
}
if (!_srs_config->get_srt_enabled(req_->vhost_)) {
bool srt_enabled = _srs_config->get_srt_enabled(req_->vhost_);
bool edge = _srs_config->get_vhost_is_edge(req_->vhost_);
if (srt_enabled && edge) {
srt_enabled = false;
srs_warn("disable SRT for edge vhost=%s", req_->vhost_.c_str());
}
if (!srt_enabled) {
return srs_error_new(ERROR_SRT_CONN, "srt disabled, vhost=%s", req_->vhost_.c_str());
}
@ -403,7 +411,13 @@ srs_error_t SrsMpegtsSrtConn::acquire_publish()
bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(req_->vhost_);
bool edge = _srs_config->get_vhost_is_edge(req_->vhost_);
if (rtc_server_enabled && rtc_enabled && !edge) {
if (rtc_enabled && edge) {
rtc_enabled = false;
srs_warn("disable WebRTC for edge vhost=%s", req_->vhost_.c_str());
}
if (rtc_server_enabled && rtc_enabled) {
if ((err = _srs_rtc_sources->fetch_or_create(req_, rtc)) != srs_success) {
return srs_error_wrap(err, "create source");
}
@ -416,11 +430,23 @@ srs_error_t SrsMpegtsSrtConn::acquire_publish()
// Bridge to RTMP and RTC streaming.
SrsSrtBridge *bridge = new SrsSrtBridge();
if (_srs_config->get_srt_to_rtmp(req_->vhost_)) {
bool srt_to_rtmp = _srs_config->get_srt_to_rtmp(req_->vhost_);
if (srt_to_rtmp && edge) {
srt_to_rtmp = false;
srs_warn("disable SRT to RTMP for edge vhost=%s", req_->vhost_.c_str());
}
if (srt_to_rtmp) {
bridge->enable_srt2rtmp(live_source);
}
if (rtc.get() && _srs_config->get_rtc_from_rtmp(req_->vhost_)) {
bool rtmp_to_rtc = _srs_config->get_rtc_from_rtmp(req_->vhost_);
if (rtmp_to_rtc && edge) {
rtmp_to_rtc = false;
srs_warn("disable RTMP to WebRTC for edge vhost=%s", req_->vhost_.c_str());
}
if (rtc.get() && rtmp_to_rtc) {
bridge->enable_srt2rtc(rtc);
}

2
trunk/src/core/srs_core_version7.hpp
View File

@ -9,6 +9,6 @@
#define VERSION_MAJOR 7
#define VERSION_MINOR 0
#define VERSION_REVISION 93
#define VERSION_REVISION 94
#endif

2
trunk/src/utest/srs_utest_app7.cpp
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@ -516,7 +516,7 @@ VOID TEST(SrsRtcPublishStreamTest, UpdateRttTypicalScenario)
// Test typical RTT update scenario for audio track
uint32_t test_ssrc = 0x87654321; // Matches audio track SSRC
int test_rtt = 50; // 50ms RTT
int test_rtt = 50; // 50ms RTT
// Call update_rtt - should find audio track and update its RTT
publish_stream->update_rtt(test_ssrc, test_rtt);