diff --git a/README.md b/README.md
index 6d2ae17..d216ad4 100644
--- a/README.md
+++ b/README.md
@@ -66,7 +66,8 @@ pip install whisperlivekit
 
 | Optional | `pip install` |
 |-----------|-------------|
-| Speaker diarization | `whisperlivekit[diarization]` |
+| Speaker diarization with Sortformer | `git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]` |
+| Speaker diarization with Diart | `diart` |
 | Original Whisper backend | `whisperlivekit[whisper]` |
 | Improved timestamps backend | `whisperlivekit[whisper-timestamped]` |
 | Apple Silicon optimization backend | `whisperlivekit[mlx-whisper]` |
@@ -185,9 +186,10 @@ The package includes an HTML/JavaScript implementation [here](https://github.com
 | Diarization options | Description | Default |
 |-----------|-------------|---------|
 | `--diarization` | Enable speaker identification | `False` |
+| `--diarization-backend` |  `diart` or `sortformer` | `diart` |
 | `--punctuation-split` | Use punctuation to improve speaker boundaries | `True` |
-| `--segmentation-model` | Hugging Face model ID for pyannote.audio segmentation model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `pyannote/segmentation-3.0` |
-| `--embedding-model` | Hugging Face model ID for pyannote.audio embedding model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `speechbrain/spkrec-ecapa-voxceleb` |
+| `--segmentation-model` | Hugging Face model ID for Diart segmentation model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `pyannote/segmentation-3.0` |
+| `--embedding-model` | Hugging Face model ID for Diart embedding model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `speechbrain/spkrec-ecapa-voxceleb` |
 
 ### 🚀 Deployment Guide
 
diff --git a/whisperlivekit/audio_processor.py b/whisperlivekit/audio_processor.py
index 9a39b3b..fcb2782 100644
--- a/whisperlivekit/audio_processor.py
+++ b/whisperlivekit/audio_processor.py
@@ -303,7 +303,7 @@ class AudioProcessor:
                 if type(item) is Silence:
                     asr_processing_logs += f" + Silence of = {item.duration:.2f}s"
                     if self.tokens:
-                        asr_processing_logs += " | last_end = {self.tokens[-1].end} |"
+                        asr_processing_logs += f" | last_end = {self.tokens[-1].end} |"
                 logger.info(asr_processing_logs)
                 
                 if type(item) is Silence:
@@ -469,7 +469,7 @@ class AudioProcessor:
                         debug_info = f"[{format_time(token.start)} : {format_time(token.end)}]"
                     if speaker != previous_speaker or not lines:
                         lines.append({
-                            "speaker": speaker,
+                            "speaker": str(speaker),
                             "text": token.text + debug_info,
                             "beg": format_time(token.start),
                             "end": format_time(token.end),
diff --git a/whisperlivekit/basic_server.py b/whisperlivekit/basic_server.py
index b49af59..ba6181e 100644
--- a/whisperlivekit/basic_server.py
+++ b/whisperlivekit/basic_server.py
@@ -52,7 +52,7 @@ async def handle_websocket_results(websocket, results_generator):
     except WebSocketDisconnect:
         logger.info("WebSocket disconnected while handling results (client likely closed connection).")
     except Exception as e:
-        logger.error(f"Error in WebSocket results handler: {e}")
+        logger.exception(f"Error in WebSocket results handler: {e}")
 
 
 @app.websocket("/asr")
diff --git a/whisperlivekit/core.py b/whisperlivekit/core.py
index a8af407..5b6052b 100644
--- a/whisperlivekit/core.py
+++ b/whisperlivekit/core.py
@@ -127,7 +127,8 @@ class TranscriptionEngine:
                     embedding_model_name=self.args.embedding_model
                 )
             elif self.args.diarization_backend == "sortformer":
-                raise ValueError('Sortformer backend in developement')
+                from whisperlivekit.diarization.sortformer_backend import SortformerDiarization
+                self.diarization = SortformerDiarization()
             else:
                 raise ValueError(f"Unknown diarization backend: {self.args.diarization_backend}")
             
diff --git a/whisperlivekit/diarization/sortformer_backend.py b/whisperlivekit/diarization/sortformer_backend.py
index 5115c91..c31e5b2 100644
--- a/whisperlivekit/diarization/sortformer_backend.py
+++ b/whisperlivekit/diarization/sortformer_backend.py
@@ -1,11 +1,492 @@
 import numpy as np
 import torch
 import logging
+import threading
+import time
+import wave
+from typing import List, Optional
+from queue import SimpleQueue, Empty
+
 from whisperlivekit.timed_objects import SpeakerSegment
 
 logger = logging.getLogger(__name__)
 
 try:
     from nemo.collections.asr.models import SortformerEncLabelModel
+    from nemo.collections.asr.modules import AudioToMelSpectrogramPreprocessor
 except ImportError:
-    raise SystemExit("""Please use `pip install "git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]"` to use the Sortformer diarization""")
\ No newline at end of file
+    raise SystemExit("""Please use `pip install "git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]"` to use the Sortformer diarization""")
+
+
+class StreamingSortformerState:
+    """
+    This class creates a class instance that will be used to store the state of the
+    streaming Sortformer model.
+
+    Attributes:
+        spkcache (torch.Tensor): Speaker cache to store embeddings from start
+        spkcache_lengths (torch.Tensor): Lengths of the speaker cache
+        spkcache_preds (torch.Tensor): The speaker predictions for the speaker cache parts
+        fifo (torch.Tensor): FIFO queue to save the embedding from the latest chunks
+        fifo_lengths (torch.Tensor): Lengths of the FIFO queue
+        fifo_preds (torch.Tensor): The speaker predictions for the FIFO queue parts
+        spk_perm (torch.Tensor): Speaker permutation information for the speaker cache
+        mean_sil_emb (torch.Tensor): Mean silence embedding
+        n_sil_frames (torch.Tensor): Number of silence frames
+    """
+
+    def __init__(self):
+        self.spkcache = None  # Speaker cache to store embeddings from start
+        self.spkcache_lengths = None
+        self.spkcache_preds = None  # speaker cache predictions
+        self.fifo = None  # to save the embedding from the latest chunks
+        self.fifo_lengths = None
+        self.fifo_preds = None
+        self.spk_perm = None
+        self.mean_sil_emb = None
+        self.n_sil_frames = None
+
+
+class SortformerDiarization:
+    def __init__(self, sample_rate: int = 16000, model_name: str = "nvidia/diar_streaming_sortformer_4spk-v2"):
+        """
+        Initialize the streaming Sortformer diarization system.
+        
+        Args:
+            sample_rate: Audio sample rate (default: 16000)
+            model_name: Pre-trained model name (default: "nvidia/diar_streaming_sortformer_4spk-v2")
+        """
+        self.sample_rate = sample_rate
+        self.speaker_segments = []
+        self.segment_lock = threading.Lock()
+        self.global_time_offset = 0.0
+        self.processed_time = 0.0
+        
+        # Load and configure the model
+        self._load_model(model_name)
+        
+        # Initialize streaming state
+        self._init_streaming_state()
+        
+        # Audio processing variables
+        self._previous_chunk_features = None
+        self._chunk_index = 0
+        self._len_prediction = None
+        
+        # Audio buffer to store PCM chunks for debugging
+        self.audio_buffer = []
+        
+        # Buffer for accumulating audio chunks until reaching chunk_duration_seconds
+        self.audio_chunk_buffer = []
+        self.accumulated_duration = 0.0
+        
+        logger.info("SortformerDiarization initialized successfully")
+
+    def _load_model(self, model_name: str):
+        """Load and configure the Sortformer model for streaming."""
+        try:
+            self.diar_model = SortformerEncLabelModel.from_pretrained(model_name)
+            self.diar_model.eval()
+
+            if torch.cuda.is_available():
+                self.diar_model.to(torch.device("cuda"))
+                logger.info("Using CUDA for Sortformer model")
+            else:
+                logger.info("Using CPU for Sortformer model")
+
+            self.diar_model.sortformer_modules.chunk_len = 10
+            self.diar_model.sortformer_modules.subsampling_factor = 10
+            self.diar_model.sortformer_modules.chunk_right_context = 0
+            self.diar_model.sortformer_modules.chunk_left_context = 10
+            self.diar_model.sortformer_modules.spkcache_len = 188
+            self.diar_model.sortformer_modules.fifo_len = 188
+            self.diar_model.sortformer_modules.spkcache_update_period = 144
+            self.diar_model.sortformer_modules.log = False
+            self.diar_model.sortformer_modules._check_streaming_parameters()
+
+            self.audio2mel = AudioToMelSpectrogramPreprocessor(
+                window_size=0.025,
+                normalize="NA",
+                n_fft=512,
+                features=128,
+                pad_to=0
+            )
+            
+            self.chunk_duration_seconds = (
+                self.diar_model.sortformer_modules.chunk_len * 
+                self.diar_model.sortformer_modules.subsampling_factor * 
+                self.diar_model.preprocessor._cfg.window_stride
+            )
+            
+            logger.info(f"Chunk duration: {self.chunk_duration_seconds:.2f}s")
+            
+        except Exception as e:
+            logger.error(f"Failed to load Sortformer model: {e}")
+            raise
+
+    def _init_streaming_state(self):
+        """Initialize the streaming state for the model."""
+        batch_size = 1
+        device = self.diar_model.device
+        
+        self.streaming_state = StreamingSortformerState()
+        self.streaming_state.spkcache = torch.zeros(
+            (batch_size, self.diar_model.sortformer_modules.spkcache_len, self.diar_model.sortformer_modules.fc_d_model), 
+            device=device
+        )
+        self.streaming_state.spkcache_preds = torch.zeros(
+            (batch_size, self.diar_model.sortformer_modules.spkcache_len, self.diar_model.sortformer_modules.n_spk), 
+            device=device
+        )
+        self.streaming_state.spkcache_lengths = torch.zeros((batch_size,), dtype=torch.long, device=device)
+        self.streaming_state.fifo = torch.zeros(
+            (batch_size, self.diar_model.sortformer_modules.fifo_len, self.diar_model.sortformer_modules.fc_d_model), 
+            device=device
+        )
+        self.streaming_state.fifo_lengths = torch.zeros((batch_size,), dtype=torch.long, device=device)
+        self.streaming_state.mean_sil_emb = torch.zeros((batch_size, self.diar_model.sortformer_modules.fc_d_model), device=device)
+        self.streaming_state.n_sil_frames = torch.zeros((batch_size,), dtype=torch.long, device=device)
+        
+        # Initialize total predictions tensor
+        self.total_preds = torch.zeros((batch_size, 0, self.diar_model.sortformer_modules.n_spk), device=device)
+
+    def insert_silence(self, silence_duration: float):
+        """
+        Insert silence period by adjusting the global time offset.
+        
+        Args:
+            silence_duration: Duration of silence in seconds
+        """
+        with self.segment_lock:
+            self.global_time_offset += silence_duration
+        logger.debug(f"Inserted silence of {silence_duration:.2f}s, new offset: {self.global_time_offset:.2f}s")
+
+    async def diarize(self, pcm_array: np.ndarray):
+        """
+        Process audio data for diarization in streaming fashion.
+        
+        Args:
+            pcm_array: Audio data as numpy array
+        """
+        try:
+            # Store PCM array for debugging
+            self.audio_buffer.append(pcm_array.copy())
+            
+            # Add to buffer and accumulate duration
+            self.audio_chunk_buffer.append(pcm_array.copy())
+            chunk_duration = len(pcm_array) / self.sample_rate
+            self.accumulated_duration += chunk_duration
+            
+            # Check if we have accumulated enough audio
+            if self.accumulated_duration < self.chunk_duration_seconds:
+                print(f"Accumulating audio: {self.accumulated_duration:.2f}/{self.chunk_duration_seconds:.2f}s")
+                return
+            
+            # Concatenate all buffered audio chunks
+            concatenated_audio = np.concatenate(self.audio_chunk_buffer)
+            
+            # Reset buffer and accumulated duration
+            self.audio_chunk_buffer = []
+            self.accumulated_duration = 0.0
+            
+            # Convert audio to torch tensor
+            audio_signal_chunk = torch.tensor(concatenated_audio).unsqueeze(0).to(self.diar_model.device)
+            audio_signal_length_chunk = torch.tensor([audio_signal_chunk.shape[1]]).to(self.diar_model.device)
+            
+            # Extract mel features
+            processed_signal_chunk, processed_signal_length_chunk = self.audio2mel.get_features(
+                audio_signal_chunk, audio_signal_length_chunk
+            )
+            
+            # Handle feature overlap for continuity
+            if self._previous_chunk_features is not None:
+                # Add overlap from previous chunk (99 frames as in offline version)
+                to_add = self._previous_chunk_features[:, :, -99:]
+                total_features = torch.concat([to_add, processed_signal_chunk], dim=2)
+            else:
+                total_features = processed_signal_chunk
+            
+            # Store current features for next iteration
+            self._previous_chunk_features = processed_signal_chunk
+            
+            # Transpose for model input
+            chunk_feat_seq_t = torch.transpose(total_features, 1, 2)
+            
+            # Process with streaming model
+            with torch.inference_mode():
+                left_offset = 8 if self._chunk_index > 0 else 0
+                right_offset = 8
+                
+                self.streaming_state, self.total_preds = self.diar_model.forward_streaming_step(
+                    processed_signal=chunk_feat_seq_t,
+                    processed_signal_length=torch.tensor([chunk_feat_seq_t.shape[1]]),
+                    streaming_state=self.streaming_state,
+                    total_preds=self.total_preds,
+                    left_offset=left_offset,
+                    right_offset=right_offset,
+                )
+                
+            # Convert predictions to speaker segments
+            self._process_predictions()
+            
+            self._chunk_index += 1
+            
+        except Exception as e:
+            logger.error(f"Error in diarize: {e}")
+            raise
+            
+        # TODO: Handle case when stream ends with partial buffer (accumulated_duration > 0 but < chunk_duration_seconds)
+
+    def _process_predictions(self):
+        """Process model predictions and convert to speaker segments."""
+        try:
+            preds_np = self.total_preds[0].cpu().numpy()
+            active_speakers = np.argmax(preds_np, axis=1)
+            
+            if self._len_prediction is None:
+                self._len_prediction = len(active_speakers)
+            
+            # Get predictions for current chunk
+            frame_duration = self.chunk_duration_seconds / self._len_prediction
+            current_chunk_preds = active_speakers[-self._len_prediction:]
+            
+            with self.segment_lock:
+                # Process predictions into segments
+                base_time = self._chunk_index * self.chunk_duration_seconds + self.global_time_offset
+                
+                for idx, spk in enumerate(current_chunk_preds):
+                    start_time = base_time + idx * frame_duration
+                    end_time = base_time + (idx + 1) * frame_duration
+                    
+                    # Check if this continues the last segment or starts a new one
+                    if (self.speaker_segments and 
+                        self.speaker_segments[-1].speaker == spk and 
+                        abs(self.speaker_segments[-1].end - start_time) < frame_duration * 0.5):
+                        # Continue existing segment
+                        self.speaker_segments[-1].end = end_time
+                    else:
+                        
+                        # Create new segment
+                        self.speaker_segments.append(SpeakerSegment(
+                            speaker=spk,
+                            start=start_time,
+                            end=end_time
+                        ))
+                        print('NEW SPEAKER, SpeakerSegment:', str(self.speaker_segments[-1]))
+                
+                # Update processed time
+                self.processed_time = max(self.processed_time, base_time + self.chunk_duration_seconds)
+                
+                logger.debug(f"Processed chunk {self._chunk_index}, total segments: {len(self.speaker_segments)}")
+                
+        except Exception as e:
+            logger.error(f"Error processing predictions: {e}")
+
+    def assign_speakers_to_tokens(self, tokens: list, use_punctuation_split: bool = False) -> list:
+        """
+        Assign speakers to tokens based on timing overlap with speaker segments.
+        
+        Args:
+            tokens: List of tokens with timing information
+            use_punctuation_split: Whether to use punctuation for boundary refinement
+            
+        Returns:
+            List of tokens with speaker assignments
+        """
+        with self.segment_lock:
+            segments = self.speaker_segments.copy()
+        
+        if not segments or not tokens:
+            logger.debug("No segments or tokens available for speaker assignment")
+            return tokens
+        
+        logger.debug(f"Assigning speakers to {len(tokens)} tokens using {len(segments)} segments")
+        if not use_punctuation_split:
+            # Simple overlap-based assignment
+            for token in tokens:
+                token.speaker = -1  # Default to no speaker
+                for segment in segments:
+                    # Check for timing overlap
+                    if not (segment.end <= token.start or segment.start >= token.end):
+                        token.speaker = segment.speaker + 1  # Convert to 1-based indexing
+                        break
+        else:
+            # Use punctuation-aware assignment (similar to diart_backend)
+            tokens = self._add_speaker_to_tokens_with_punctuation(segments, tokens)
+        
+        return tokens
+
+    def _add_speaker_to_tokens_with_punctuation(self, segments: List[SpeakerSegment], tokens: list) -> list:
+        """
+        Assign speakers to tokens with punctuation-aware boundary adjustment.
+        
+        Args:
+            segments: List of speaker segments
+            tokens: List of tokens to assign speakers to
+            
+        Returns:
+            List of tokens with speaker assignments
+        """
+        punctuation_marks = {'.', '!', '?'}
+        punctuation_tokens = [token for token in tokens if token.text.strip() in punctuation_marks]
+        
+        # Convert segments to concatenated format
+        segments_concatenated = self._concatenate_speakers(segments)
+        
+        # Adjust segment boundaries based on punctuation
+        for ind, segment in enumerate(segments_concatenated):
+            for i, punctuation_token in enumerate(punctuation_tokens):
+                if punctuation_token.start > segment['end']:
+                    after_length = punctuation_token.start - segment['end']
+                    before_length = segment['end'] - punctuation_tokens[i - 1].end if i > 0 else float('inf')
+                    
+                    if before_length > after_length:
+                        segment['end'] = punctuation_token.start
+                        if i < len(punctuation_tokens) - 1 and ind + 1 < len(segments_concatenated):
+                            segments_concatenated[ind + 1]['begin'] = punctuation_token.start
+                    else:
+                        segment['end'] = punctuation_tokens[i - 1].end if i > 0 else segment['end']
+                        if i < len(punctuation_tokens) - 1 and ind - 1 >= 0:
+                            segments_concatenated[ind - 1]['begin'] = punctuation_tokens[i - 1].end
+                    break
+        
+        # Ensure non-overlapping tokens
+        last_end = 0.0
+        for token in tokens:
+            start = max(last_end + 0.01, token.start)
+            token.start = start
+            token.end = max(start, token.end)
+            last_end = token.end
+        
+        # Assign speakers based on adjusted segments
+        ind_last_speaker = 0
+        for segment in segments_concatenated:
+            for i, token in enumerate(tokens[ind_last_speaker:]):
+                if token.end <= segment['end']:
+                    token.speaker = segment['speaker']
+                    ind_last_speaker = i + 1
+                elif token.start > segment['end']:
+                    break
+        
+        return tokens
+
+    def _concatenate_speakers(self, segments: List[SpeakerSegment]) -> List[dict]:
+        """
+        Concatenate consecutive segments from the same speaker.
+        
+        Args:
+            segments: List of speaker segments
+            
+        Returns:
+            List of concatenated speaker segments
+        """
+        if not segments:
+            return []
+            
+        segments_concatenated = [{"speaker": segments[0].speaker + 1, "begin": segments[0].start, "end": segments[0].end}]
+        
+        for segment in segments[1:]:
+            speaker = segment.speaker + 1
+            if segments_concatenated[-1]['speaker'] != speaker:
+                segments_concatenated.append({"speaker": speaker, "begin": segment.start, "end": segment.end})
+            else:
+                segments_concatenated[-1]['end'] = segment.end
+                
+        return segments_concatenated
+
+    def get_segments(self) -> List[SpeakerSegment]:
+        """Get a copy of the current speaker segments."""
+        with self.segment_lock:
+            return self.speaker_segments.copy()
+
+    def clear_old_segments(self, older_than: float = 30.0):
+        """Clear segments older than the specified time."""
+        with self.segment_lock:
+            current_time = self.processed_time
+            self.speaker_segments = [
+                segment for segment in self.speaker_segments 
+                if current_time - segment.end < older_than
+            ]
+            logger.debug(f"Cleared old segments, remaining: {len(self.speaker_segments)}")
+
+    def close(self):
+        """Close the diarization system and clean up resources."""
+        logger.info("Closing SortformerDiarization")
+        with self.segment_lock:
+            self.speaker_segments.clear()
+        
+        # Save audio buffer to WAV file for debugging
+        if self.audio_buffer:
+            try:
+                # Concatenate all PCM chunks
+                concatenated_audio = np.concatenate(self.audio_buffer)
+                
+                # Convert from float32 back to int16 for WAV file
+                audio_data_int16 = (concatenated_audio * 32767).astype(np.int16)
+                
+                # Write to WAV file
+                with wave.open("diarization_audio.wav", "wb") as wav_file:
+                    wav_file.setnchannels(1)  # Mono audio
+                    wav_file.setsampwidth(2)   # 2 bytes per sample (int16)
+                    wav_file.setframerate(self.sample_rate)
+                    wav_file.writeframes(audio_data_int16.tobytes())
+                
+                logger.info(f"Saved {len(concatenated_audio)} samples to diarization_audio.wav")
+            except Exception as e:
+                logger.error(f"Error saving audio to WAV file: {e}")
+        else:
+            logger.info("No audio data to save")
+
+
+def extract_number(s: str) -> int:
+    """Extract number from speaker string (compatibility function)."""
+    import re
+    m = re.search(r'\d+', s)
+    return int(m.group()) if m else 0
+
+
+if __name__ == '__main__':
+    import asyncio
+    import librosa
+    
+    async def main():
+        """Main function to test SortformerDiarization with the same example as offline version."""
+        # Load and prepare audio (same as offline version)
+        try:
+            an4_audio = 'audio_test.mp3'
+            signal, sr = librosa.load(an4_audio, sr=16000)
+            signal = signal[:16000*30]  # 30 seconds
+        except Exception as e:
+            print(f"Error loading audio file: {e}")
+            return
+        
+        print("\n" + "=" * 50)
+        print("Expected ground truth:")
+        print("Speaker 0: 0:00 - 0:09")
+        print("Speaker 1: 0:09 - 0:19") 
+        print("Speaker 2: 0:19 - 0:25")
+        print("Speaker 0: 0:25 - 0:30")
+        print("=" * 50)
+        
+        # Create diarization instance
+        diarization = SortformerDiarization(sample_rate=16000)
+        
+        # Chunk and process audio
+        chunk_size = 16000  # 1 second
+        print(f"Processing {len(signal)} samples in {len(signal) // chunk_size} chunks...")
+        
+        for i in range(0, len(signal), chunk_size):
+            chunk = signal[i:i+chunk_size]
+            await diarization.diarize(chunk)
+            print(f"Processed chunk {i // chunk_size + 1}")
+        
+        # Close and save WAV
+        # diarization.close()
+        
+        # Print results
+        segments = diarization.get_segments()
+        print("\nDiarization results:")
+        for segment in segments:
+            print(f"Speaker {segment.speaker}: {segment.start:.2f}s - {segment.end:.2f}s")
+    
+    asyncio.run(main())