diff --git a/src/channel/meida_channel/video_channel_send.h b/src/channel/meida_channel/video_channel_send.h
index f3334d5..c0c15b4 100644
--- a/src/channel/meida_channel/video_channel_send.h
+++ b/src/channel/meida_channel/video_channel_send.h
@@ -36,6 +36,12 @@ class VideoChannelSend {
   std::vector<std::unique_ptr<RtpPacket>> GeneratePadding(
       uint32_t payload_size, int64_t capture_timestamp_ms);
 
+  int64_t GetTransportSeqAndIncrement() {
+    int64_t transport_seq = rtp_video_sender_->GetTransportSequenceNumber();
+    rtp_video_sender_->IncrementTransportSequenceNumber();
+    return transport_seq;
+  }
+
  public:
   void Initialize(rtp::PAYLOAD_TYPE payload_type);
   void Destroy();
diff --git a/src/channel/rtp_channel/rtp_video_sender.h b/src/channel/rtp_channel/rtp_video_sender.h
index 12a50ab..32bed4f 100644
--- a/src/channel/rtp_channel/rtp_video_sender.h
+++ b/src/channel/rtp_channel/rtp_video_sender.h
@@ -35,6 +35,10 @@ class RtpVideoSender : public ThreadBase {
   uint32_t GetSsrc() { return ssrc_; }
   void OnReceiverReport(const ReceiverReport &receiver_report);
 
+  int64_t GetTransportSequenceNumber() { return transport_seq_; }
+
+  void IncrementTransportSequenceNumber() { transport_seq_++; }
+
  private:
   int SendRtpPacket(
       std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send);
diff --git a/src/common/any_invocable.h b/src/common/any_invocable.h
new file mode 100644
index 0000000..b393ba6
--- /dev/null
+++ b/src/common/any_invocable.h
@@ -0,0 +1,69 @@
+/*
+ * @Author: DI JUNKUN
+ * @Date: 2025-03-14
+ * Copyright (c) 2025 by DI JUNKUN, All Rights Reserved.
+ */
+
+#ifndef _ANY_INVOCABLE_H_
+#define _ANY_INVOCABLE_H_
+
+#include <functional>
+#include <iostream>
+#include <memory>
+
+// 简化版的 AnyInvocable
+template <typename Signature>
+class AnyInvocable;
+
+template <typename R, typename... Args>
+class AnyInvocable<R(Args...)> {
+ public:
+  // 默认构造函数
+  AnyInvocable() = default;
+
+  // 构造函数：接受一个可以调用的对象
+  template <typename Callable>
+  AnyInvocable(Callable&& callable)
+      : callable_(std::make_unique<CallableWrapper<Callable>>(
+            std::forward<Callable>(callable))) {}
+
+  // 调用运算符
+  R operator()(Args... args) {
+    return callable_->Invoke(std::forward<Args>(args)...);
+  }
+
+  // 移动构造函数
+  AnyInvocable(AnyInvocable&&) = default;
+  // 移动赋值运算符
+  AnyInvocable& operator=(AnyInvocable&&) = default;
+
+ private:
+  // 抽象基类，允许不同类型的可调用对象
+  struct CallableBase {
+    virtual ~CallableBase() = default;
+    virtual R Invoke(Args&&... args) = 0;
+  };
+
+  // 模板派生类：实际存储 callable 对象
+  template <typename Callable>
+  struct CallableWrapper : public CallableBase {
+    CallableWrapper(Callable&& callable)
+        : callable_(std::forward<Callable>(callable)) {}
+
+    R Invoke(Args&&... args) override {
+      return callable_(std::forward<Args>(args)...);
+    }
+
+    Callable callable_;
+  };
+
+  std::unique_ptr<CallableBase> callable_;
+};
+
+// 简单的包装函数
+template <typename R, typename... Args>
+AnyInvocable<R(Args...)> MakeMoveOnlyFunction(std::function<R(Args...)>&& f) {
+  return AnyInvocable<R(Args...)>(std::move(f));
+}
+
+#endif
\ No newline at end of file
diff --git a/src/common/task_queue.h b/src/common/task_queue.h
new file mode 100644
index 0000000..1736650
--- /dev/null
+++ b/src/common/task_queue.h
@@ -0,0 +1,107 @@
+/*
+ * @Author: DI JUNKUN
+ * @Date: 2025-03-14
+ * Copyright (c) 2025 by DI JUNKUN, All Rights Reserved.
+ */
+
+#ifndef _TASK_QUEUE_H_
+#define _TASK_QUEUE_H_
+
+#include <chrono>
+#include <condition_variable>
+#include <functional>
+#include <memory>
+#include <mutex>
+#include <queue>
+#include <thread>
+#include <vector>
+
+#include "any_invocable.h"
+
+class TaskQueue {
+ public:
+  TaskQueue(size_t numThreads = 1)
+      : stop_(false), workers_(), taskQueue_(), mutex_(), cond_var_() {
+    for (size_t i = 0; i < numThreads; i++) {
+      workers_.emplace_back([this]() { this->WorkerThread(); });
+    }
+  }
+
+  ~TaskQueue() {
+    {
+      std::unique_lock<std::mutex> lock(mutex_);
+      stop_ = true;
+    }
+    cond_var_.notify_all();
+    for (std::thread &worker : workers_) {
+      if (worker.joinable()) {
+        worker.join();
+      }
+    }
+  }
+
+  // 立即执行任务
+  void PostTask(AnyInvocable<void()> task) {
+    PostDelayedTask(std::move(task), 0);
+  }
+
+  // 延迟执行任务
+  void PostDelayedTask(AnyInvocable<void()> task, int delay_ms) {
+    auto execute_time =
+        std::chrono::steady_clock::now() + std::chrono::milliseconds(delay_ms);
+    {
+      std::unique_lock<std::mutex> lock(mutex_);
+      taskQueue_.emplace(execute_time,
+                         std::move(task));  // 确保参数匹配 TaskItem 构造
+    }
+    cond_var_.notify_one();
+  }
+
+ private:
+  struct TaskItem {
+    std::chrono::steady_clock::time_point execute_time;
+    AnyInvocable<void()> task = nullptr;
+
+    TaskItem(std::chrono::steady_clock::time_point time,
+             AnyInvocable<void()> func)
+        : execute_time(time), task(std::move(func)) {}
+
+    bool operator>(const TaskItem &other) const {
+      return execute_time > other.execute_time;
+    }
+  };
+
+  void WorkerThread() {
+    while (true) {
+      AnyInvocable<void()> task;
+      {
+        std::unique_lock<std::mutex> lock(mutex_);
+        cond_var_.wait(lock, [this]() { return !taskQueue_.empty() || stop_; });
+
+        if (stop_ && taskQueue_.empty()) return;
+
+        auto now = std::chrono::steady_clock::now();
+        if (taskQueue_.top().execute_time > now) {
+          cond_var_.wait_until(lock, taskQueue_.top().execute_time,
+                               [this]() { return stop_; });
+        }
+
+        if (stop_ && taskQueue_.empty()) return;
+
+        task = std::move(
+            const_cast<AnyInvocable<void()> &>(taskQueue_.top().task));
+        taskQueue_.pop();
+      }
+      task();  // 执行任务
+    }
+  }
+
+  std::vector<std::thread> workers_;
+  std::priority_queue<TaskItem, std::vector<TaskItem>, std::greater<>>
+      taskQueue_;
+  mutable std::mutex mutex_;
+  std::condition_variable cond_var_;
+  bool stop_;
+};
+
+#endif
diff --git a/src/qos/pacing_controller.cc b/src/qos/pacing_controller.cc
index 5a105c9..594ed49 100644
--- a/src/qos/pacing_controller.cc
+++ b/src/qos/pacing_controller.cc
@@ -412,6 +412,7 @@ void PacingController::ProcessPackets() {
   PacedPacketInfo pacing_info;
   DataSize recommended_probe_size = DataSize::Zero();
   bool is_probing = prober_.is_probing();
+  LOG_WARN("is probing");
   if (is_probing) {
     // Probe timing is sensitive, and handled explicitly by BitrateProber, so
     // use actual send time rather than target.
@@ -433,7 +434,8 @@ void PacingController::ProcessPackets() {
     // exhausted.
     std::unique_ptr<RtpPacketToSend> rtp_packet =
         GetPendingPacket(pacing_info, target_send_time, now);
-    if (rtp_packet == nullptr) {
+    if (!rtp_packet) {
+      LOG_WARN("rtp_packet == nullptr");
       // No packet available to send, check if we should send padding.
       if (now - target_send_time > kMaxPaddingReplayDuration) {
         // The target send time is more than `kMaxPaddingReplayDuration` behind
@@ -472,6 +474,7 @@ void PacingController::ProcessPackets() {
                        transport_overhead_per_packet_;
       }
 
+      LOG_ERROR("Send packet_size {}", rtp_packet->Size());
       packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
       for (auto& packet : packet_sender_->FetchFec()) {
         EnqueuePacket(std::move(packet));
diff --git a/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp b/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
index fe1cd30..eb47462 100644
--- a/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
+++ b/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
@@ -246,55 +246,63 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildPadding(
     bool use_rtp_packet_to_send) {
   std::vector<std::unique_ptr<RtpPacket>> rtp_packets;
 
-  version_ = kRtpVersion;
-  has_padding_ = true;
-  has_extension_ = true;
-  csrc_count_ = 0;
-  marker_ = 0;
-  uint8_t payload_type = rtp::PAYLOAD_TYPE(payload_type_ - 1);
-  sequence_number_++;
-  timestamp_ = kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_ms);
+  uint32_t remaining_size = payload_size;
+  while (remaining_size > 0) {
+    uint32_t current_payload_size =
+        std::min<uint32_t>(remaining_size, MAX_NALU_LEN);
 
-  rtp_packet_frame_.clear();
-  rtp_packet_frame_.push_back((version_ << 6) | (has_padding_ << 5) |
-                              (has_extension_ << 4) | csrc_count_);
-  rtp_packet_frame_.push_back((marker_ << 7) | payload_type);
-  rtp_packet_frame_.push_back((sequence_number_ >> 8) & 0xFF);
-  rtp_packet_frame_.push_back(sequence_number_ & 0xFF);
-  rtp_packet_frame_.push_back((timestamp_ >> 24) & 0xFF);
-  rtp_packet_frame_.push_back((timestamp_ >> 16) & 0xFF);
-  rtp_packet_frame_.push_back((timestamp_ >> 8) & 0xFF);
-  rtp_packet_frame_.push_back(timestamp_ & 0xFF);
-  rtp_packet_frame_.push_back((ssrc_ >> 24) & 0xFF);
-  rtp_packet_frame_.push_back((ssrc_ >> 16) & 0xFF);
-  rtp_packet_frame_.push_back((ssrc_ >> 8) & 0xFF);
-  rtp_packet_frame_.push_back(ssrc_ & 0xFF);
+    version_ = kRtpVersion;
+    has_padding_ = true;
+    has_extension_ = true;
+    csrc_count_ = 0;
+    marker_ = 0;
+    uint8_t payload_type = rtp::PAYLOAD_TYPE(payload_type_ - 1);
+    sequence_number_++;
+    timestamp_ =
+        kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_ms);
 
-  for (uint32_t index = 0; index < csrc_count_ && !csrcs_.empty(); index++) {
-    rtp_packet_frame_.push_back((csrcs_[index] >> 24) & 0xFF);
-    rtp_packet_frame_.push_back((csrcs_[index] >> 16) & 0xFF);
-    rtp_packet_frame_.push_back((csrcs_[index] >> 8) & 0xFF);
-    rtp_packet_frame_.push_back(csrcs_[index] & 0xFF);
-  }
+    rtp_packet_frame_.clear();
+    rtp_packet_frame_.push_back((version_ << 6) | (has_padding_ << 5) |
+                                (has_extension_ << 4) | csrc_count_);
+    rtp_packet_frame_.push_back((marker_ << 7) | payload_type);
+    rtp_packet_frame_.push_back((sequence_number_ >> 8) & 0xFF);
+    rtp_packet_frame_.push_back(sequence_number_ & 0xFF);
+    rtp_packet_frame_.push_back((timestamp_ >> 24) & 0xFF);
+    rtp_packet_frame_.push_back((timestamp_ >> 16) & 0xFF);
+    rtp_packet_frame_.push_back((timestamp_ >> 8) & 0xFF);
+    rtp_packet_frame_.push_back(timestamp_ & 0xFF);
+    rtp_packet_frame_.push_back((ssrc_ >> 24) & 0xFF);
+    rtp_packet_frame_.push_back((ssrc_ >> 16) & 0xFF);
+    rtp_packet_frame_.push_back((ssrc_ >> 8) & 0xFF);
+    rtp_packet_frame_.push_back(ssrc_ & 0xFF);
 
-  if (has_extension_) {
-    AddAbsSendTimeExtension(rtp_packet_frame_);
-  }
+    for (uint32_t index = 0; index < csrc_count_ && !csrcs_.empty(); index++) {
+      rtp_packet_frame_.push_back((csrcs_[index] >> 24) & 0xFF);
+      rtp_packet_frame_.push_back((csrcs_[index] >> 16) & 0xFF);
+      rtp_packet_frame_.push_back((csrcs_[index] >> 8) & 0xFF);
+      rtp_packet_frame_.push_back(csrcs_[index] & 0xFF);
+    }
 
-  // Add padding bytes
-  uint32_t padding_size = payload_size;
-  rtp_packet_frame_.insert(rtp_packet_frame_.end(), padding_size - 1, 0);
-  rtp_packet_frame_.push_back(padding_size);
+    if (has_extension_) {
+      AddAbsSendTimeExtension(rtp_packet_frame_);
+    }
 
-  if (use_rtp_packet_to_send) {
-    std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet =
-        std::make_unique<webrtc::RtpPacketToSend>();
-    rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
-    rtp_packets.emplace_back(std::move(rtp_packet));
-  } else {
-    std::unique_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
-    rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
-    rtp_packets.emplace_back(std::move(rtp_packet));
+    // Add padding bytes
+    uint32_t padding_size = current_payload_size;
+    rtp_packet_frame_.insert(rtp_packet_frame_.end(), padding_size - 1, 0);
+
+    if (use_rtp_packet_to_send) {
+      std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet =
+          std::make_unique<webrtc::RtpPacketToSend>();
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packets.emplace_back(std::move(rtp_packet));
+    } else {
+      std::unique_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packets.emplace_back(std::move(rtp_packet));
+    }
+
+    remaining_size -= current_payload_size;
   }
 
   return rtp_packets;
diff --git a/src/transport/packet_sender.cpp b/src/transport/packet_sender.cpp
index 8a4561e..9c2b6e9 100644
--- a/src/transport/packet_sender.cpp
+++ b/src/transport/packet_sender.cpp
@@ -25,11 +25,16 @@ PacketSender::GeneratePadding(webrtc::DataSize size) {
   std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> to_send_rtp_packets;
   std::vector<std::unique_ptr<RtpPacket>> rtp_packets =
       generat_padding_func_(size.bytes(), clock_->CurrentTime().ms());
-  // for (auto &packet : rtp_packets) {
-  //   std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send(
-  //       static_cast<webrtc::RtpPacketToSend *>(packet.release()));
-  //   to_send_rtp_packets.push_back(std::move(rtp_packet_to_send));
-  // }
+  for (auto &packet : rtp_packets) {
+    std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send(
+        static_cast<webrtc::RtpPacketToSend *>(packet.release()));
+
+    rtp_packet_to_send->set_capture_time(clock_->CurrentTime());
+    rtp_packet_to_send->set_transport_sequence_number((transport_seq_)++);
+    rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kPadding);
+
+    to_send_rtp_packets.push_back(std::move(rtp_packet_to_send));
+  }
 
   return to_send_rtp_packets;
 }
@@ -67,23 +72,24 @@ void PacketSender::SetPacingRates(webrtc::DataRate pacing_rate,
 
 void PacketSender::EnqueuePackets(
     std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> packets) {
-  // task_queue_->PostTask()
-  for (auto &packet : packets) {
-    size_t packet_size = packet->payload_size() + packet->padding_size();
-    if (include_overhead_) {
-      packet_size += packet->headers_size();
+  task_queue_.PostTask([this, packets = std::move(packets)]() mutable {
+    for (auto &packet : packets) {
+      size_t packet_size = packet->payload_size() + packet->padding_size();
+      if (include_overhead_) {
+        packet_size += packet->headers_size();
+      }
+      packet_size_.Apply(1, packet_size);
+      pacing_controller_.EnqueuePacket(std::move(packet));
     }
-    packet_size_.Apply(1, packet_size);
-    pacing_controller_.EnqueuePacket(std::move(packet));
-  }
-  MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
+    MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
+  });
 }
 
 void PacketSender::RemovePacketsForSsrc(uint32_t ssrc) {
-  // task_queue_->PostTask(SafeTask(safety_.flag(), [this, ssrc] {
-  pacing_controller_.RemovePacketsForSsrc(ssrc);
-  MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
-  // }));
+  task_queue_.PostTask([this, ssrc] {
+    pacing_controller_.RemovePacketsForSsrc(ssrc);
+    MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
+  });
 }
 
 void PacketSender::SetAccountForAudioPackets(bool account_for_audio) {
diff --git a/src/transport/packet_sender.h b/src/transport/packet_sender.h
index 69761ba..a3da1d6 100644
--- a/src/transport/packet_sender.h
+++ b/src/transport/packet_sender.h
@@ -21,6 +21,7 @@
 #include "rtc_base/numerics/exp_filter.h"
 #include "rtp_packet_pacer.h"
 #include "rtp_packet_to_send.h"
+#include "task_queue.h"
 
 class PacketSender : public webrtc::RtpPacketPacer,
                      public webrtc::PacingController::PacketSender {
@@ -197,6 +198,9 @@ class PacketSender : public webrtc::RtpPacketPacer,
   Stats current_stats_;
   // Protects against ProcessPackets reentry from packet sent receipts.
   bool processing_packets_ = false;
+
+  TaskQueue task_queue_;
+  int64_t transport_seq_ = 0;
 };
 
 #endif
\ No newline at end of file