diff --git a/demos/streaming_asr_server/conf/ws_ds2_application.yaml b/demos/streaming_asr_server/conf/ws_ds2_application.yaml
index e7ce59c2602a6a88bcaefc6fd8741e30646b238d..f3f60b968dda563a9702647efae5b079e6b1a8fa 100644
--- a/demos/streaming_asr_server/conf/ws_ds2_application.yaml
+++ b/demos/streaming_asr_server/conf/ws_ds2_application.yaml
@@ -45,7 +45,7 @@ asr_online-inference:
         sample_width: 2
         window_n: 7     # frame
         shift_n: 4      # frame
-        window_ms: 20   # ms
+        window_ms: 25   # ms
         shift_ms: 10    # ms
 
 
@@ -80,5 +80,5 @@ asr_online-onnx:
         sample_width: 2
         window_n: 7     # frame
         shift_n: 4      # frame
-        window_ms: 20   # ms
+        window_ms: 25   # ms
         shift_ms: 10    # ms
diff --git a/demos/streaming_asr_server/local/websocket_client.py b/demos/streaming_asr_server/local/websocket_client.py
index 0371240251252816d126099c448d64a662baaff0..51ae7a2f45591b60c28bd77d611401995682b909 100644
--- a/demos/streaming_asr_server/local/websocket_client.py
+++ b/demos/streaming_asr_server/local/websocket_client.py
@@ -13,6 +13,8 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 # calc avg RTF(NOT Accurate): grep -rn RTF log.txt | awk '{print $NF}' | awk -F "=" '{sum += $NF} END {print "all time",sum, "audio num", NR,  "RTF", sum/NR}'
+# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --punc.server_ip 127.0.0.1 --punc.port 8190 --wavfile ./zh.wav
+# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --wavfile ./zh.wav
 import argparse
 import asyncio
 import codecs
diff --git a/demos/streaming_asr_server/test.sh b/demos/streaming_asr_server/test.sh
index f09068d47308251bc9db48413eae92ff923cdaaf..67a5ec4c5023fc49d4585b608cabb87d88255811 100755
--- a/demos/streaming_asr_server/test.sh
+++ b/demos/streaming_asr_server/test.sh
@@ -3,11 +3,9 @@ wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav
 
 # read the wav and pass it to only streaming asr service
 # If `127.0.0.1` is not accessible, you need to use the actual service IP address.
-# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --wavfile ./zh.wav
 paddlespeech_client asr_online --server_ip 127.0.0.1 --port 8090 --input ./zh.wav
 
 # read the wav and call streaming and punc service
 # If `127.0.0.1` is not accessible, you need to use the actual service IP address.
-# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --punc.server_ip 127.0.0.1 --punc.port 8190 --wavfile ./zh.wav
 paddlespeech_client asr_online --server_ip 127.0.0.1 --port 8290 --punc.server_ip 127.0.0.1 --punc.port 8190 --input ./zh.wav
 
diff --git a/paddlespeech/server/conf/ws_ds2_application.yaml b/paddlespeech/server/conf/ws_ds2_application.yaml
index e7ce59c2602a6a88bcaefc6fd8741e30646b238d..f3f60b968dda563a9702647efae5b079e6b1a8fa 100644
--- a/paddlespeech/server/conf/ws_ds2_application.yaml
+++ b/paddlespeech/server/conf/ws_ds2_application.yaml
@@ -45,7 +45,7 @@ asr_online-inference:
         sample_width: 2
         window_n: 7     # frame
         shift_n: 4      # frame
-        window_ms: 20   # ms
+        window_ms: 25   # ms
         shift_ms: 10    # ms
 
 
@@ -80,5 +80,5 @@ asr_online-onnx:
         sample_width: 2
         window_n: 7     # frame
         shift_n: 4      # frame
-        window_ms: 20   # ms
+        window_ms: 25   # ms
         shift_ms: 10    # ms