diff --git a/src/forward/constant.rs b/src/forward/constant.rs
index 88297564..323b1389 100644
--- a/src/forward/constant.rs
+++ b/src/forward/constant.rs
@@ -1,2 +1,2 @@
 pub(crate) const VIDEO_KIND: &str = "video";
-pub(crate) const AUDIO_KIND: &str = "audio";
\ No newline at end of file
+pub(crate) const AUDIO_KIND: &str = "audio";
diff --git a/src/forward/forward_internal.rs b/src/forward/forward_internal.rs
index e2230f9d..aa355c19 100644
--- a/src/forward/forward_internal.rs
+++ b/src/forward/forward_internal.rs
@@ -7,9 +7,9 @@ use anyhow::Result;
 use log::info;
 use tokio::sync::mpsc::{unbounded_channel, UnboundedSender};
 use tokio::sync::RwLock;
-use webrtc::api::APIBuilder;
 use webrtc::api::interceptor_registry::register_default_interceptors;
 use webrtc::api::media_engine::MediaEngine;
+use webrtc::api::APIBuilder;
 use webrtc::ice_transport::ice_candidate::RTCIceCandidateInit;
 use webrtc::ice_transport::ice_server::RTCIceServer;
 use webrtc::interceptor::registry::Registry;
@@ -18,13 +18,13 @@ use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
 use webrtc::peer_connection::RTCPeerConnection;
 use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication;
 use webrtc::rtp::packet::Packet;
-use webrtc::rtp_transceiver::RTCRtpTransceiverInit;
 use webrtc::rtp_transceiver::rtp_codec::{
     RTCRtpCodecCapability, RTCRtpCodecParameters, RTPCodecType,
 };
 use webrtc::rtp_transceiver::rtp_transceiver_direction::RTCRtpTransceiverDirection;
-use webrtc::track::track_local::{TrackLocal, TrackLocalWriter};
+use webrtc::rtp_transceiver::RTCRtpTransceiverInit;
 use webrtc::track::track_local::track_local_static_rtp::TrackLocalStaticRTP;
+use webrtc::track::track_local::{TrackLocal, TrackLocalWriter};
 use webrtc::track::track_remote::TrackRemote;
 
 use super::constant::*;
diff --git a/src/forward/mod.rs b/src/forward/mod.rs
index a594691d..7c3c82a6 100644
--- a/src/forward/mod.rs
+++ b/src/forward/mod.rs
@@ -7,8 +7,8 @@ use tokio::sync::Mutex;
 use webrtc::ice_transport::ice_candidate::RTCIceCandidateInit;
 use webrtc::ice_transport::ice_server::RTCIceServer;
 use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
-use webrtc::peer_connection::RTCPeerConnection;
 use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
+use webrtc::peer_connection::RTCPeerConnection;
 use webrtc::sdp::{MediaDescription, SessionDescription};
 
 use constant::*;