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

trunk/conf/edge.conf

@@ -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;
+    }
 }

trunk/doc/CHANGELOG.md

@@ -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)

trunk/src/app/srs_app_rtc_api.cpp

@@ -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());
     }

trunk/src/app/srs_app_rtc_conn.cpp

@@ -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.

trunk/src/app/srs_app_rtmp_conn.cpp

@@ -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

trunk/src/app/srs_app_rtmp_source.cpp

@@ -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();
+    srs_error_t err = srs_success;
-    if (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;

trunk/src/app/srs_app_srt_conn.cpp

@@ -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);
     }
 

trunk/src/core/srs_core_version7.hpp

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

trunk/src/utest/srs_utest_app7.cpp

@@ -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);