AI: WebRTC: Support optional msid attribute per RFC 8830. v7.0.126 (ossrs#4570)

Author: ossrs-ai

.vscode/README.md

@@ -40,6 +40,11 @@ cmake --build $HOME/git/srs/trunk/cmake/build
 
 ## macOS: SRS UTest
 
+The most straightforward way is to select a test name like `WorkflowRtcManuallyVerifyForPublisher`,
+then select `Debug gtest (macOS CodeLLDB)` and run the debug.
+
+Or you can use the following way to run specified test from the test panel.
+
 Install the following extensions:
 
 - C++ TestMate

.vscode/launch.json

@@ -2,7 +2,7 @@
     "version": "0.2.0",
     "configurations": [
         {
-          "name": "Launch SRS with conf/console.conf",
+          "name": "Debug SRS with conf/console.conf",
           "type": "cppdbg",
           "request": "launch",
           "program": "${workspaceFolder}/trunk/cmake/build/srs",
@@ -30,7 +30,7 @@
           }
         },
         {
-          "name": "Launch SRS with conf/rtc.conf",
+          "name": "Debug SRS with conf/rtc.conf",
           "type": "cppdbg",
           "request": "launch",
           "program": "${workspaceFolder}/trunk/cmake/build/srs",
@@ -58,40 +58,41 @@
           }
         },
         {
-          "name": "Launch SRS with console.conf",
-          "type": "cppdbg",
+          "name": "Debug srs-proxy",
+          "type": "go",
+          "request": "launch",
+          "mode": "auto",
+          "cwd": "${workspaceFolder}/proxy",
+          "program": "${workspaceFolder}/proxy"        
+        },
+        {
+          "name": "Debug SRS (macOS, CodeLLDB) console.conf",
+          "type": "lldb",
           "request": "launch",
           "program": "${workspaceFolder}/trunk/cmake/build/srs",
           "args": ["-c", "console.conf"],
-          "stopAtEntry": false,
           "cwd": "${workspaceFolder}/trunk",
-          "environment": [],
-          "externalConsole": false,
-          "linux": {
-            "MIMode": "gdb"
-          },
-          "osx": {
-            "MIMode": "lldb"
-          },
-          "setupCommands": [
-            {
-              "description": "Enable pretty-printing for gdb",
-              "text": "-enable-pretty-printing",
-              "ignoreFailures": true
-            }
+          "stopOnEntry": false,
+          "terminal": "integrated",
+          "initCommands": [
+            "command script import lldb.formatters.cpp.libcxx"
           ],
           "preLaunchTask": "build",
-          "logging": {
-            "engineLogging": true
-          }
+          "env": {},
+          "sourceLanguages": ["cpp"]
         },
         {
-          "name": "Launch srs-proxy",
-          "type": "go",
+          "name": "Debug gtest (macOS CodeLLDB)",
+          "type": "lldb",
           "request": "launch",
-          "mode": "auto",
-          "cwd": "${workspaceFolder}/proxy",
-          "program": "${workspaceFolder}/proxy"        
-        }    
+          "program": "${workspaceFolder}/trunk/cmake/build/utest",
+          "args": ["--gtest_filter=*${selectedText}*"],
+          "cwd": "${workspaceFolder}/trunk",
+          "terminal": "integrated",
+          "initCommands": [
+            "command script import lldb.formatters.cpp.libcxx"
+          ],
+          "sourceLanguages": ["cpp"]
+        }
     ]
 }

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v7-changes"></a>
 
 ## SRS 7.0 Changelog
+* v7.0, 2025-11-11, 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)
 * v7.0, 2025-11-08, AI: WebRTC: Support VP9 codec for WebRTC-to-WebRTC streaming. v7.0.123 (#4548)

trunk/src/app/srs_app_rtc_conn.cpp

@@ -3601,12 +3601,25 @@ srs_error_t SrsRtcPublisherNegotiator::negotiate_publish_capability(SrsRtcUserCo
         for (int j = 0; j < (int)remote_media_desc.ssrc_infos_.size(); ++j) {
             const SrsSSRCInfo &ssrc_info = remote_media_desc.ssrc_infos_.at(j);
 
+            // Generate msid because it's optional in sdp.
+            string msid_tracker = ssrc_info.msid_tracker_;
+            if (msid_tracker.empty()) {
+                msid_tracker = srs_fmt_sprintf("track-%s-%s-%d", 
+                    track_desc->type_.c_str(), ssrc_info.cname_.c_str(), ssrc_info.ssrc_);
+            }
+
+            // Generate msid because it's optional in sdp.
+            string msid = ssrc_info.msid_;
+            if (msid.empty()) {
+                msid = req->app_ + "/" + req->stream_;
+            }
+
             // ssrc have same track id, will be description in the same track description.
-            if (track_id != ssrc_info.msid_tracker_) {
+            if (track_id != msid_tracker) {
                 SrsRtcTrackDescription *track_desc_copy = track_desc->copy();
                 track_desc_copy->ssrc_ = ssrc_info.ssrc_;
-                track_desc_copy->id_ = ssrc_info.msid_tracker_;
-                track_desc_copy->msid_ = ssrc_info.msid_;
+                track_desc_copy->id_ = msid_tracker;
+                track_desc_copy->msid_ = msid;
 
                 if (remote_media_desc.is_audio() && !stream_desc->audio_track_desc_) {
                     stream_desc->audio_track_desc_ = track_desc_copy;
@@ -3616,7 +3629,7 @@ srs_error_t SrsRtcPublisherNegotiator::negotiate_publish_capability(SrsRtcUserCo
                     srs_freep(track_desc_copy);
                 }
             }
-            track_id = ssrc_info.msid_tracker_;
+            track_id = msid_tracker;
         }
 
         // set track fec_ssrc and rtx_ssrc

trunk/src/core/srs_core_version7.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR 7
 #define VERSION_MINOR 0
-#define VERSION_REVISION 125
+#define VERSION_REVISION 126
 
 #endif

trunk/src/protocol/srs_protocol_sdp.cpp

@@ -375,9 +375,10 @@ vector<SrsMediaPayloadType> SrsMediaDesc::find_media_with_encoding_name(const st
     transform(encoding_name.begin(), encoding_name.end(), upper_name.begin(), ::toupper);
 
     for (size_t i = 0; i < payload_types_.size(); ++i) {
-        if (payload_types_[i].encoding_name_ == std::string(lower_name.c_str()) ||
-            payload_types_[i].encoding_name_ == std::string(upper_name.c_str())) {
-            payloads.push_back(payload_types_[i]);
+        SrsMediaPayloadType payload = payload_types_[i];
+        if (payload.encoding_name_ == std::string(lower_name.c_str()) ||
+            payload.encoding_name_ == std::string(upper_name.c_str())) {
+            payloads.push_back(payload);
         }
     }
 

trunk/src/utest/srs_utest_ai12.cpp

@@ -2039,6 +2039,116 @@ VOID TEST(SrsRtcPublisherNegotiatorTest, TypicalUseScenario)
     EXPECT_EQ("video", video_sdp.media_descs_[0].type_);
 }
 
+VOID TEST(SrsRtcPublisherNegotiatorTest, LibdatachannelUseScenario)
+{
+    srs_error_t err;
+
+    // Create SrsRtcPublisherNegotiator
+    SrsUniquePtr<SrsRtcPublisherNegotiator> negotiator(new SrsRtcPublisherNegotiator());
+
+    // Create mock request for initialization
+    SrsUniquePtr<MockRtcConnectionRequest> mock_request(new MockRtcConnectionRequest("test.vhost", "live", "stream1"));
+
+    // Create mock RTC user config with remote SDP
+    SrsUniquePtr<SrsRtcUserConfig> ruc(new SrsRtcUserConfig());
+    ruc->req_ = mock_request->copy();
+    ruc->publish_ = true;
+    ruc->dtls_ = true;
+    ruc->srtp_ = true;
+    ruc->audio_before_video_ = true;
+
+    // SDP from issue 4570 - libdatachannel format with video first, then audio
+    ruc->remote_sdp_str_ =
+        "v=0\r\n"
+        "o=- rtc 4158491451 0 IN IP4 127.0.0.1\r\n"
+        "s=-\r\n"
+        "t=0 0\r\n"
+        "a=group:BUNDLE video audio\r\n"
+        "a=group:LS video audio\r\n"
+        "a=msid-semantic:WMS *\r\n"
+        "a=ice-options:ice2,trickle\r\n"
+        "a=fingerprint:sha-256 28:37:F7:18:77:FC:46:33:6F:B2:0F:12:83:C2:BF:5C:61:5E:96:EB:4B:B9:97:81:92:7C:82:10:97:B8:8E:60\r\n"
+        "m=video 56144 UDP/TLS/RTP/SAVPF 96 97\r\n"
+        "c=IN IP4 172.24.64.1\r\n"
+        "a=mid:video\r\n"
+        "a=sendonly\r\n"
+        "a=ssrc:42 cname:video-send\r\n"
+        "a=rtcp-mux\r\n"
+        "a=rtpmap:96 H264/90000\r\n"
+        "a=rtcp-fb:96 nack\r\n"
+        "a=rtcp-fb:96 nack pli\r\n"
+        "a=rtcp-fb:96 goog-remb\r\n"
+        "a=fmtp:96 profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1\r\n"
+        "a=rtpmap:97 RTX/90000\r\n"
+        "a=fmtp:97 apt=96\r\n"
+        "a=setup:actpass\r\n"
+        "a=ice-ufrag:fEw/\r\n"
+        "a=ice-pwd:jBua8YGWQKc/Vn6Y9EZ9+0\r\n"
+        "a=candidate:1 1 UDP 2122317823 172.24.64.1 56144 typ host\r\n"
+        "a=candidate:2 1 UDP 2122315767 10.0.0.94 56144 typ host\r\n"
+        "a=candidate:3 1 UDP 1686189695 111.43.134.137 56144 typ srflx raddr 0.0.0.0 rport 0\r\n"
+        "a=end-of-candidates\r\n"
+        "m=audio 56144 UDP/TLS/RTP/SAVPF 111\r\n"
+        "c=IN IP4 172.24.64.1\r\n"
+        "a=mid:audio\r\n"
+        "a=sendonly\r\n"
+        "a=ssrc:43 cname:audio-send\r\n"
+        "a=rtcp-mux\r\n"
+        "a=rtpmap:111 opus/48000/2\r\n"
+        "a=fmtp:111 minptime=10;maxaveragebitrate=98000;stereo=1;sprop-stereo=1;useinbandfec=1\r\n"
+        "a=setup:actpass\r\n"
+        "a=ice-ufrag:fEw/\r\n"
+        "a=ice-pwd:jBua8YGWQKc/Vn6Y9EZ9+0\r\n";
+
+    // Parse the remote SDP
+    HELPER_EXPECT_SUCCESS(ruc->remote_sdp_.parse(ruc->remote_sdp_str_));
+
+    // Create stream description for negotiation output
+    SrsUniquePtr<SrsRtcSourceDescription> stream_desc(new SrsRtcSourceDescription());
+
+    // Test negotiate_publish_capability - typical WebRTC publisher negotiation
+    HELPER_EXPECT_SUCCESS(negotiator->negotiate_publish_capability(ruc.get(), stream_desc.get()));
+
+    // Verify that stream description was populated with audio and video tracks
+    EXPECT_TRUE(stream_desc->audio_track_desc_ != NULL);
+    EXPECT_FALSE(stream_desc->video_track_descs_.empty());
+    EXPECT_EQ("audio", stream_desc->audio_track_desc_->type_);
+    EXPECT_EQ("video", stream_desc->video_track_descs_[0]->type_);
+
+    // Test generate_publish_local_sdp - create answer SDP
+    SrsSdp local_sdp;
+    HELPER_EXPECT_SUCCESS(negotiator->generate_publish_local_sdp(
+        ruc->req_, local_sdp, stream_desc.get(),
+        ruc->remote_sdp_.is_unified(), ruc->audio_before_video_));
+
+    // Verify that local SDP was generated with media descriptions
+    EXPECT_FALSE(local_sdp.media_descs_.empty());
+
+    // Find audio and video media descriptions
+    bool has_audio = false, has_video = false;
+    for (size_t i = 0; i < local_sdp.media_descs_.size(); i++) {
+        if (local_sdp.media_descs_[i].type_ == "audio")
+            has_audio = true;
+        if (local_sdp.media_descs_[i].type_ == "video")
+            has_video = true;
+    }
+    EXPECT_TRUE(has_audio);
+    EXPECT_TRUE(has_video);
+
+    // Test individual SDP generation methods
+    SrsSdp audio_sdp, video_sdp;
+    HELPER_EXPECT_SUCCESS(negotiator->generate_publish_local_sdp_for_audio(audio_sdp, stream_desc.get()));
+    HELPER_EXPECT_SUCCESS(negotiator->generate_publish_local_sdp_for_video(video_sdp, stream_desc.get(), true));
+
+    // Verify audio SDP generation
+    EXPECT_FALSE(audio_sdp.media_descs_.empty());
+    EXPECT_EQ("audio", audio_sdp.media_descs_[0].type_);
+
+    // Verify video SDP generation
+    EXPECT_FALSE(video_sdp.media_descs_.empty());
+    EXPECT_EQ("video", video_sdp.media_descs_[0].type_);
+}
+
 VOID TEST(SrsRtcConnectionTest, InitializeTypicalScenario)
 {
     srs_error_t err;