Some webclient fixes

server/src/client/SpeakingClient.cpp

@@ -716,6 +716,7 @@ void SpeakingClient::processJoin() {
     } else if(this->getType() == ClientType::CLIENT_WEB) {
         this->rtc_client_id = this->server->rtc_server().create_rtp_client(dynamic_pointer_cast<SpeakingClient>(this->ref()));
     }
+    this->rtc_session_pending_describe = true;
 
     TIMING_STEP(timings, "server reg ");
     ref_server->getGroupManager()->cleanupAssignments(this->getClientDatabaseId());
@@ -929,7 +930,11 @@ command_result SpeakingClient::handleCommand(Command &command) {
 
 command_result SpeakingClient::handleCommandRtcSessionDescribe(Command &command) {
     CMD_REQ_SERVER;
+    if(this->rtc_session_pending_describe) {
+        this->rtc_session_pending_describe = false;
+    } else {
         CMD_CHK_AND_INC_FLOOD_POINTS(15);
+    }
 
     uint32_t mode;
     if(command["mode"].string() == "offer") {

server/src/client/SpeakingClient.h

@@ -97,6 +97,7 @@ namespace ts::server {
                 std::shared_ptr<Json::Value> identityData;
             } handshake;
 
+            bool rtc_session_pending_describe{false};
             rtc::RTCClientId rtc_client_id{0};
     };
 }

server/src/client/web/WebClient.cpp

@@ -414,7 +414,8 @@ void WebClient::disconnectFinal() {
 
     this->processLeave();
 
-    this->ssl_handler.finalize();
+    /* We do not finalize here since we might still try to send some data */
+    /* this->ssl_handler.finalize(); */
     this->handle->unregisterConnection(static_pointer_cast<WebClient>(self_lock));
 }
 

server/src/rtc/imports.h

@@ -9,6 +9,7 @@ extern "C" {
 struct NativeCallbacks {
     uint32_t version;
 
+    void(*log)(uint8_t /* level */, const void* /* callback data */, const char* /* message */, uint32_t /* length */);
     void(*free_client_data)(const void*);
 
     void(*client_stream_assignment)(const void* /* callback data */, uint32_t /* stream id */, const void* /* source callback data */);

server/src/rtc/lib.cpp

@@ -17,6 +17,47 @@ struct LibCallbackData {
     std::weak_ptr<SpeakingClient> weak_ref;
 };
 
+#define log(...) \
+switch(level) { \
+    case 0: \
+        logTrace(__VA_ARGS__); \
+        break; \
+    case 1: \
+        debugMessage(__VA_ARGS__); \
+        break; \
+    case 2: \
+        logMessage(__VA_ARGS__); \
+        break; \
+    case 3: \
+        logWarning(__VA_ARGS__); \
+        break; \
+    case 4: \
+        logError(__VA_ARGS__); \
+        break; \
+    case 5: \
+        logCritical(__VA_ARGS__); \
+        break; \
+    default: \
+        debugMessage(__VA_ARGS__); \
+        break; \
+}
+
+void librtc_callback_log(uint8_t level, const void* callback_data_ptr, const char* message_ptr, uint32_t length) {
+    auto callback_data = (LibCallbackData*) callback_data_ptr;
+    std::string_view message{message_ptr, length};
+
+    if(callback_data) {
+        auto source_client = callback_data->weak_ref.lock();
+        if(!source_client) { return; }
+
+        log(source_client->getServerId(), "{} [WebRTC] {}", CLIENT_STR_LOG_PREFIX_(source_client), message);
+    } else {
+        log(LOG_GENERAL, "{}", message);
+    }
+}
+
+#undef log
+
 void librtc_callback_free_client_data(const void* data) {
     delete (LibCallbackData*) data;
 }
@@ -56,6 +97,8 @@ void librtc_callback_client_offer_generated(const void* callback_data_ptr, const
     notify.put_unchecked(0, "mode", "offer");
     notify.put_unchecked(0, "sdp", std::string_view{offer, offer_length});
     target_client->sendCommand(notify);
+    /* don't blame the client if we require him to do anything */
+    target_client->rtc_session_pending_describe = true;
 }
 
 void librtc_callback_client_audio_start(const void* callback_data_ptr, uint32_t stream_id, const void* source_data_ptr) {
@@ -137,6 +180,7 @@ void librtc_callback_client_audio_sender_data(const void* callback_data_ptr, con
 static NativeCallbacks native_callbacks{
         .version = 1,
 
+        .log = librtc_callback_log,
         .free_client_data = librtc_callback_free_client_data,
 
         .client_stream_assignment = librtc_callback_client_stream_assignment,