AI: Fix race condition causing immediate deletion of new sources. v7.0.127 (#4449)

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v7-changes"></a>
 
 ## SRS 7.0 Changelog
+* v7.0, 2025-11-14, AI: Fix race condition causing immediate deletion of new sources. v7.0.127 (#4449)
 * v7.0, 2025-11-11, AI: WebRTC: Support optional msid attribute per RFC 8830. v7.0.126 (#4570)
 * v7.0, 2025-11-11, AI: SRT: Stop TS parsing after codec detection. v7.0.125 (#4569)
 * v7.0, 2025-11-09, AI: WebRTC: Support G.711 (PCMU/PCMA) audio codec for WebRTC. v7.0.124 (#4075)

trunk/research/players/js/srs.sdk.js

@@ -34,6 +34,9 @@ function SrsRtcWhipWhepAsync() {
     self.displayStream = null;
     self.userStream = null;
 
+    // Store the WHIP session resource URL from Location header for cleanup.
+    self.resourceUrl = null;
+
     // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
     // @url The WebRTC url to publish with, for example:
     //      http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
@@ -111,6 +114,14 @@ function SrsRtcWhipWhepAsync() {
                 if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                 const data = xhr.responseText;
                 console.log("Got answer: ", data);
+
+                // Extract Location header for WHIP session resource URL.
+                const location = xhr.getResponseHeader('Location');
+                if (location) {
+                    self.resourceUrl = new URL(location, url).href;
+                    console.log(`WHIP session resource URL: ${self.resourceUrl}`);
+                }
+
                 return data.code ? reject(xhr) : resolve(data);
             }
             xhr.open('POST', url, true);
@@ -159,6 +170,14 @@ function SrsRtcWhipWhepAsync() {
                 if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                 const data = xhr.responseText;
                 console.log("Got answer: ", data);
+
+                // Extract Location header for WHEP session resource URL.
+                const location = xhr.getResponseHeader('Location');
+                if (location) {
+                    self.resourceUrl = new URL(location, url).href;
+                    console.log(`WHEP session resource URL: ${self.resourceUrl}`);
+                }
+
                 return data.code ? reject(xhr) : resolve(data);
             }
             xhr.open('POST', url, true);
@@ -190,6 +209,25 @@ function SrsRtcWhipWhepAsync() {
             });
             self.userStream = null;
         }
+
+        // Send DELETE request to WHIP session resource URL to cleanup server resources.
+        if (self.resourceUrl) {
+            const xhr = new XMLHttpRequest();
+            xhr.open('DELETE', self.resourceUrl, true);
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status === 200) {
+                    console.log(`WHIP session deleted: ${self.resourceUrl}`);
+                } else {
+                    console.warn(`Failed to delete WHIP session: ${self.resourceUrl}, status: ${xhr.status}`);
+                }
+            };
+            xhr.onerror = function() {
+                console.warn(`Error deleting WHIP session: ${self.resourceUrl}`);
+            };
+            xhr.send();
+            self.resourceUrl = null;
+        }
     };
 
     // The callback when got local stream.

trunk/src/app/srs_app_rtc_api.cpp

@@ -491,16 +491,26 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter *w, ISrsHtt
         vcodec = r->query_get("codec");
     }
     string acodec = r->query_get("acodec");
+    // For client to specifies whether encrypt by SRTP.
+    string srtp = r->query_get("encrypt");
+    string dtls = r->query_get("dtls");
 
-    srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, vcodec=%s, acodec=%s",
+    srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, vcodec=%s, acodec=%s, srtp=%s, dtls=%s",
               streamurl.c_str(), api.c_str(), tid.c_str(), clientip.c_str(), ruc.req_->app_.c_str(), ruc.req_->stream_.c_str(),
-              remote_sdp_str.length(), eip.c_str(), vcodec.c_str(), acodec.c_str());
+              remote_sdp_str.length(), eip.c_str(), vcodec.c_str(), acodec.c_str(), srtp.c_str(), dtls.c_str());
 
     ruc.eip_ = eip;
     ruc.vcodec_ = vcodec;
     ruc.acodec_ = acodec;
     ruc.publish_ = true;
-    ruc.dtls_ = ruc.srtp_ = true;
+
+    // For client to specifies whether encrypt by SRTP.
+    ruc.dtls_ = (dtls != "false");
+    if (srtp.empty()) {
+        ruc.srtp_ = config_->get_rtc_server_encrypt();
+    } else {
+        ruc.srtp_ = (srtp != "false");
+    }
 
     // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
     ruc.remote_sdp_str_ = remote_sdp_str;

trunk/src/app/srs_app_rtc_source.cpp

@@ -335,7 +335,7 @@ srs_error_t SrsRtcSourceManager::fetch_or_create(ISrsRequest *r, SrsSharedPtr<Sr
             pps = source;
         } else {
             SrsSharedPtr<SrsRtcSource> source = SrsSharedPtr<SrsRtcSource>(new SrsRtcSource());
-            srs_trace("new rtc source, stream_url=%s", stream_url.c_str());
+            srs_trace("new rtc source, stream_url=%s, dead=%d", stream_url.c_str(), source->stream_is_dead());
             pps = source;
 
             pool_[stream_url] = source;
@@ -399,7 +399,10 @@ SrsRtcSource::SrsRtcSource()
     circuit_breaker_ = _srs_circuit_breaker;
 
     pli_for_rtmp_ = pli_elapsed_ = 0;
-    stream_die_at_ = 0;
+    // Initialize stream_die_at_ to current time to prevent newly created sources
+    // from being immediately considered dead by stream_is_dead() check.
+    // @see https://github.com/ossrs/srs/issues/4449
+    stream_die_at_ = srs_time_now_cached();
 
     app_factory_ = _srs_app_factory;
 }

trunk/src/app/srs_app_rtmp_source.cpp

@@ -1659,7 +1659,7 @@ srs_error_t SrsLiveSourceManager::fetch_or_create(ISrsRequest *r, SrsSharedPtr<S
             pps = source;
         } else {
             SrsSharedPtr<SrsLiveSource> source = app_factory_->create_live_source();
-            srs_trace("new live source, stream_url=%s", stream_url.c_str());
+            srs_trace("new live source, stream_url=%s, dead=%d", stream_url.c_str(), source->stream_is_dead());
             pps = source;
 
             // Callback to notify request of source creation
@@ -1781,7 +1781,10 @@ SrsLiveSource::SrsLiveSource()
     mix_queue_ = new SrsMixQueue();
 
     can_publish_ = true;
-    stream_die_at_ = 0;
+    // Initialize stream_die_at_ to current time to prevent newly created sources
+    // from being immediately considered dead by stream_is_dead() check.
+    // @see https://github.com/ossrs/srs/issues/4449
+    stream_die_at_ = srs_time_now_cached();
     publisher_idle_at_ = 0;
 
     rtmp_bridge_ = NULL;

trunk/src/app/srs_app_rtsp_source.cpp

@@ -194,7 +194,7 @@ srs_error_t SrsRtspSourceManager::fetch_or_create(ISrsRequest *r, SrsSharedPtr<S
             pps = source;
         } else {
             SrsSharedPtr<SrsRtspSource> source = SrsSharedPtr<SrsRtspSource>(new SrsRtspSource());
-            srs_trace("new rtsp source, stream_url=%s", stream_url.c_str());
+            srs_trace("new rtsp source, stream_url=%s, dead=%d", stream_url.c_str(), source->stream_is_dead());
             pps = source;
 
             pool_[stream_url] = source;
@@ -249,7 +249,10 @@ SrsRtspSource::SrsRtspSource()
 
     req_ = NULL;
 
-    stream_die_at_ = 0;
+    // Initialize stream_die_at_ to current time to prevent newly created sources
+    // from being immediately considered dead by stream_is_dead() check.
+    // @see https://github.com/ossrs/srs/issues/4449
+    stream_die_at_ = srs_time_now_cached();
 
     stat_ = _srs_stat;
     circuit_breaker_ = _srs_circuit_breaker;

trunk/src/app/srs_app_srt_source.cpp

@@ -179,7 +179,7 @@ srs_error_t SrsSrtSourceManager::fetch_or_create(ISrsRequest *r, SrsSharedPtr<Sr
             pps = source;
         } else {
             SrsSharedPtr<SrsSrtSource> source(new SrsSrtSource());
-            srs_trace("new srt source, stream_url=%s", stream_url.c_str());
+            srs_trace("new srt source, stream_url=%s, dead=%d", stream_url.c_str(), source->stream_is_dead());
             pps = source;
 
             pool_[stream_url] = source;
@@ -1230,7 +1230,10 @@ SrsSrtSource::SrsSrtSource()
     req_ = NULL;
     can_publish_ = true;
     srt_bridge_ = NULL;
-    stream_die_at_ = 0;
+    // Initialize stream_die_at_ to current time to prevent newly created sources
+    // from being immediately considered dead by stream_is_dead() check.
+    // @see https://github.com/ossrs/srs/issues/4449
+    stream_die_at_ = srs_time_now_cached();
 
     stat_ = _srs_stat;
     format_ = new SrsSrtFormat();

trunk/src/core/srs_core_version7.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR 7
 #define VERSION_MINOR 0
-#define VERSION_REVISION 126
+#define VERSION_REVISION 127
 
 #endif