diff --git a/whisperlivekit/audio_processor.py b/whisperlivekit/audio_processor.py
index 8e23ef2..9a39b3b 100644
--- a/whisperlivekit/audio_processor.py
+++ b/whisperlivekit/audio_processor.py
@@ -301,7 +301,9 @@ class AudioProcessor:
                 transcription_lag_s = max(0.0, time() - self.beg_loop - self.end_buffer)
                 asr_processing_logs = f"internal_buffer={asr_internal_buffer_duration_s:.2f}s | lag={transcription_lag_s:.2f}s |"
                 if type(item) is Silence:
-                    asr_processing_logs += f" + Silence of = {item.duration:.2f}s | last_end = {self.tokens[-1].end} |"
+                    asr_processing_logs += f" + Silence of = {item.duration:.2f}s"
+                    if self.tokens:
+                        asr_processing_logs += " | last_end = {self.tokens[-1].end} |"
                 logger.info(asr_processing_logs)
                 
                 if type(item) is Silence:
@@ -445,12 +447,15 @@ class AudioProcessor:
                 last_end_diarized = 0
                 undiarized_text = []
                 current_time = time() - self.beg_loop if self.beg_loop else None
-                tokens, buffer_transcription = handle_silences(tokens, buffer_transcription, current_time, self.silence)
+                tokens, buffer_transcription, buffer_diarization = handle_silences(tokens, buffer_transcription, buffer_diarization, current_time, self.silence)
                 for token in tokens:
                     speaker = token.speaker
                     
+                    if speaker == -1: #Speaker -1 means no attributed by diarization. In the frontend, it should appear under 'Speaker 1'
+                        speaker = 1
+                    
                     # Handle diarization
-                    if self.args.diarization:
+                    if self.args.diarization and not tokens[-1].speaker == -2:
                         if (speaker in [-1, 0]) and token.end >= end_attributed_speaker:
                             undiarized_text.append(token.text)
                             continue
diff --git a/whisperlivekit/remove_silences.py b/whisperlivekit/remove_silences.py
index 1ea7f6d..5091f00 100644
--- a/whisperlivekit/remove_silences.py
+++ b/whisperlivekit/remove_silences.py
@@ -77,9 +77,9 @@ def no_token_to_silence(tokens):
             new_tokens.append(token)
     return new_tokens
             
-def ends_with_silence(tokens, buffer_transcription, current_time, vac_detected_silence):
+def ends_with_silence(tokens, buffer_transcription, buffer_diarization, current_time, vac_detected_silence):
     if not tokens:
-        return []
+        return [], buffer_transcription, buffer_diarization
     last_token = tokens[-1]
     if tokens and (
         current_time - last_token.end >= END_SILENCE_DURATION 
@@ -97,13 +97,14 @@ def ends_with_silence(tokens, buffer_transcription, current_time, vac_detected_s
                     probability=0.95
                 )
             )
-        #We validate the buffer has because of the silence
-    return tokens
+        buffer_transcription = "" # for whisperstreaming backend, we should probably validate the buffer has because of the silence
+        buffer_diarization  = ""
+    return tokens, buffer_transcription, buffer_diarization
     
 
-def handle_silences(tokens, buffer_transcription, current_time, vac_detected_silence):
+def handle_silences(tokens, buffer_transcription, buffer_diarization, current_time, vac_detected_silence):
     tokens = blank_to_silence(tokens) #useful for simulstreaming backend which tends to generate [BLANK_AUDIO] text
     tokens = no_token_to_silence(tokens)
-    tokens = ends_with_silence(tokens, buffer_transcription, current_time, vac_detected_silence)
-    return tokens, buffer_transcription
+    tokens, buffer_transcription, buffer_diarization = ends_with_silence(tokens, buffer_transcription, buffer_diarization, current_time, vac_detected_silence)
+    return tokens, buffer_transcription, buffer_diarization
      
\ No newline at end of file
diff --git a/whisperlivekit/web/live_transcription.js b/whisperlivekit/web/live_transcription.js
index 8661219..f2efb18 100644
--- a/whisperlivekit/web/live_transcription.js
+++ b/whisperlivekit/web/live_transcription.js
@@ -3,7 +3,7 @@
 let isRecording = false;
 let websocket = null;
 let recorder = null;
-let chunkDuration = 1000;
+let chunkDuration = 100;
 let websocketUrl = "ws://localhost:8000/asr";
 let userClosing = false;
 let wakeLock = null;
@@ -269,9 +269,7 @@ function renderLinesWithBuffer(
         speakerLabel = `<span class='loading'><span class="spinner"></span><span id='timeInfo'><span class="loading-diarization-value">${fmt1(
           remaining_time_diarization
         )}</span> second(s) of audio are undergoing diarization</span></span>`;
-      } else if (item.speaker == -1) {
-        speakerLabel = `<span id="speaker">Speaker 1<span id='timeInfo'>${timeInfo}</span></span>`;
-      } else if (item.speaker !== -1 && item.speaker !== 0) {
+      } else if (item.speaker !== 0) {
         speakerLabel = `<span id="speaker">Speaker ${item.speaker}<span id='timeInfo'>${timeInfo}</span></span>`;
       }