From 75607d0723ee7f01d2c0e7c87870c46299c885db Mon Sep 17 00:00:00 2001
From: Xiaoniu <nmgchenhaibo@foxmail.com>
Date: Wed, 19 May 2021 20:21:20 +0800
Subject: [PATCH 1/3] Fix #2364: Live: Fail for stream busy if GB28181 is
 publishing.

---
 trunk/src/app/srs_app_rtmp_conn.cpp | 9 +++++++++
 1 file changed, 9 insertions(+)

diff --git a/trunk/src/app/srs_app_rtmp_conn.cpp b/trunk/src/app/srs_app_rtmp_conn.cpp
index 29947eaa2..1f1b256cb 100644
--- a/trunk/src/app/srs_app_rtmp_conn.cpp
+++ b/trunk/src/app/srs_app_rtmp_conn.cpp
@@ -960,6 +960,15 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsLiveSource* source)
     srs_error_t err = srs_success;
     
     SrsRequest* req = info->req;
+	
+	// @see https://github.com/ossrs/srs/issues/2364
+    // Check whether GB28181 stream is busy.
+#if defined(SRS_GB28181)
+    SrsGb28181RtmpMuxer* gb28181 = _srs_gb28181->fetch_rtmpmuxer(req->stream);
+    if (gb28181 != NULL) {
+        return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "gb28181 stream %s busy", req->get_stream_url().c_str());
+    }
+#endif
 
     // Check whether RTC stream is busy.
 #ifdef SRS_RTC

From ef617b5fc69fed613f403928ab2e83ebed824e74 Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Wed, 19 May 2021 20:26:35 +0800
Subject: [PATCH 2/3] Demo: Refine H5 demo page

---
 trunk/research/api-server/static-dir/index.html | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/trunk/research/api-server/static-dir/index.html b/trunk/research/api-server/static-dir/index.html
index b3ecfc979..e73a6e75b 100755
--- a/trunk/research/api-server/static-dir/index.html
+++ b/trunk/research/api-server/static-dir/index.html
@@ -11,8 +11,8 @@
     点击进入<a id="cn" href="#">SRS控制台</a>
 </p>
 <p>
-    Click <a href="players/">here</a> to start SRS player.<br/>
-    点击进入<a href="players/">SRS播放器</a>
+    Click <a href="players/?autostart=true">here</a> to start SRS player.<br/>
+    点击进入<a href="players/?autostart=true">SRS播放器</a>
 </p>
 <p><a href="https://github.com/ossrs/srs">SRS Team &copy; 2021</a></p>
 <script type="text/javascript">

From d55af6be44fc196823fd7379aedba42cff1bb9b0 Mon Sep 17 00:00:00 2001
From: root <root@localhost.localdomain>
Date: Tue, 18 May 2021 15:57:22 +0800
Subject: [PATCH 3/3] Fix #2362: Allow WebRTC to play before publishing, for
 GB28181 as such. 4.0.117

---
 README.md                            |   2 +
 trunk/src/app/srs_app_rtc_source.cpp | 102 ++++++++++++++++-----------
 trunk/src/app/srs_app_rtc_source.hpp |   3 +
 trunk/src/core/srs_core_version4.hpp |   2 +-
 4 files changed, 68 insertions(+), 41 deletions(-)

diff --git a/README.md b/README.md
index 28c5187a0..8b3abdbb3 100755
--- a/README.md
+++ b/README.md
@@ -176,6 +176,7 @@ The ports used by SRS:
 
 ## V4 changes
 
+* v4.0, 2021-05-19, Fix [#2362][bug #2362]: Allow WebRTC to play before publishing, for GB28181 as such. 4.0.117
 * v4.0, 2021-05-18, Fix [#2355][bug #2355]: GB28181: Fix play by RTC bug. 4.0.116
 * v4.0, 2021-05-15, SRT: Build SRT from source by SRS. 4.0.115
 * v4.0, 2021-05-15, Rename SrsConsumer* to SrsLiveConsumer*. 4.0.114
@@ -1926,6 +1927,7 @@ Winlin
 [bug #2304]: https://github.com/ossrs/srs/issues/2304#issuecomment-826009290
 [bug #2355]: https://github.com/ossrs/srs/issues/2355
 [bug #307]: https://github.com/ossrs/srs/issues/307
+[bug #2362]: https://github.com/ossrs/srs/issues/2362
 [bug #yyyyyyyyyyyyy]: https://github.com/ossrs/srs/issues/yyyyyyyyyyyyy
 
 [bug #1631]: https://github.com/ossrs/srs/issues/1631
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index 3ee15606c..4fc594256 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -367,7 +367,58 @@ srs_error_t SrsRtcSource::initialize(SrsRequest* r)
 
     req = r->copy();
 
-    return err;
+	// Create default relations to allow play before publishing.
+	// @see https://github.com/ossrs/srs/issues/2362
+	init_for_play_before_publishing();
+
+	return err;
+}
+
+void SrsRtcSource::init_for_play_before_publishing()
+{
+    // If the stream description has already been setup by RTC publisher,
+    // we should ignore and it's ok, because we only need to setup it for bridger.
+    if (stream_desc_) {
+        return;
+    }
+
+    SrsRtcSourceDescription* stream_desc = new SrsRtcSourceDescription();
+    SrsAutoFree(SrsRtcSourceDescription, stream_desc);
+
+    // audio track description
+    if (true) {
+        SrsRtcTrackDescription* audio_track_desc = new SrsRtcTrackDescription();
+        stream_desc->audio_track_desc_ = audio_track_desc;
+
+        audio_track_desc->type_ = "audio";
+        audio_track_desc->id_ = "audio-" + srs_random_str(8);
+
+        uint32_t audio_ssrc = SrsRtcSSRCGenerator::instance()->generate_ssrc();
+        audio_track_desc->ssrc_ = audio_ssrc;
+        audio_track_desc->direction_ = "recvonly";
+
+        audio_track_desc->media_ = new SrsAudioPayload(kAudioPayloadType, "opus", kAudioSamplerate, kAudioChannel);
+    }
+
+    // video track description
+    if (true) {
+        SrsRtcTrackDescription* video_track_desc = new SrsRtcTrackDescription();
+        stream_desc->video_track_descs_.push_back(video_track_desc);
+
+        video_track_desc->type_ = "video";
+        video_track_desc->id_ = "video-" + srs_random_str(8);
+
+        uint32_t video_ssrc = SrsRtcSSRCGenerator::instance()->generate_ssrc();
+        video_track_desc->ssrc_ = video_ssrc;
+        video_track_desc->direction_ = "recvonly";
+
+        SrsVideoPayload* video_payload = new SrsVideoPayload(kVideoPayloadType, "H264", kVideoSamplerate);
+        video_track_desc->media_ = video_payload;
+
+        video_payload->set_h264_param_desc("level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f");
+    }
+
+    set_stream_desc(stream_desc);
 }
 
 void SrsRtcSource::update_auth(SrsRequest* r)
@@ -542,9 +593,6 @@ void SrsRtcSource::on_unpublish()
         srs_freep(bridger_);
     }
 
-    // release unpublish stream description.
-    set_stream_desc(NULL);
-
     // TODO: FIXME: Handle by statistic.
 }
 
@@ -675,46 +723,20 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
     audio_sequence = 0;
     video_sequence = 0;
 
-    SrsRtcSourceDescription* stream_desc = new SrsRtcSourceDescription();
-    SrsAutoFree(SrsRtcSourceDescription, stream_desc);
-
-    // audio track description
+    // audio track ssrc
     if (true) {
-        SrsRtcTrackDescription* audio_track_desc = new SrsRtcTrackDescription();
-        stream_desc->audio_track_desc_ = audio_track_desc;
-
-        audio_track_desc->type_ = "audio";
-        audio_track_desc->id_ = "audio-"  + srs_random_str(8);
-
-        audio_ssrc = SrsRtcSSRCGenerator::instance()->generate_ssrc();
-        audio_track_desc->ssrc_ = audio_ssrc;
-        audio_track_desc->direction_ = "recvonly";
-
-        audio_track_desc->media_ = new SrsAudioPayload(kAudioPayloadType, "opus", kAudioSamplerate, kAudioChannel);
+        std::vector<SrsRtcTrackDescription*> descs = source->get_track_desc("audio", "opus");
+        if (!descs.empty()) {
+            audio_ssrc = descs.at(0)->ssrc_;
+        }
     }
 
-    // video track description
+    // video track ssrc
     if (true) {
-        SrsRtcTrackDescription* video_track_desc = new SrsRtcTrackDescription();
-        stream_desc->video_track_descs_.push_back(video_track_desc);
-
-        video_track_desc->type_ = "video";
-        video_track_desc->id_ = "video-"  + srs_random_str(8);
-
-        video_ssrc = SrsRtcSSRCGenerator::instance()->generate_ssrc();
-        video_track_desc->ssrc_ = video_ssrc;
-        video_track_desc->direction_ = "recvonly";
-
-        SrsVideoPayload* video_payload = new SrsVideoPayload(kVideoPayloadType, "H264", kVideoSamplerate);
-        video_track_desc->media_ = video_payload;
-
-        video_payload->set_h264_param_desc("level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f");
-    }
-
-    // If the stream description has already been setup by RTC publisher,
-    // we should ignore and it's ok, because we only need to setup it for bridger.
-    if (!source_->has_stream_desc()) {
-        source_->set_stream_desc(stream_desc);
+        std::vector<SrsRtcTrackDescription*> descs = source->get_track_desc("video", "H264");
+        if (!descs.empty()) {
+            video_ssrc = descs.at(0)->ssrc_;
+        }
     }
 }
 
diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
index 23fae5b34..00b63490d 100644
--- a/trunk/src/app/srs_app_rtc_source.hpp
+++ b/trunk/src/app/srs_app_rtc_source.hpp
@@ -209,6 +209,9 @@ public:
     virtual ~SrsRtcSource();
 public:
     virtual srs_error_t initialize(SrsRequest* r);
+private:
+    void init_for_play_before_publishing();
+public:
     // Update the authentication information in request.
     virtual void update_auth(SrsRequest* r);
 private:
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index 31be6ba1a..9cac0d463 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    116
+#define VERSION_REVISION    117
 
 #endif