diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index 8825df2..11c299c 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -156,7 +156,7 @@
 }
 
 RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
-    std::map<uint32_t, RtpState> suspended_ssrcs,
+    const std::map<uint32_t, RtpState>& suspended_ssrcs,
     const std::map<uint32_t, RtpPayloadState>& states,
     const RtpConfig& rtp_config,
     int rtcp_report_interval_ms,
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index ac4213d..62af78c 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -65,7 +65,7 @@
 
   // TODO(tommi): Change to std::unique_ptr<>.
   RtpVideoSenderInterface* CreateRtpVideoSender(
-      std::map<uint32_t, RtpState> suspended_ssrcs,
+      const std::map<uint32_t, RtpState>& suspended_ssrcs,
       const std::map<uint32_t, RtpPayloadState>&
           states,  // move states into RtpTransportControllerSend
       const RtpConfig& rtp_config,
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index 2aa6d73..f68c4bf 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -96,7 +96,7 @@
   virtual PacketRouter* packet_router() = 0;
 
   virtual RtpVideoSenderInterface* CreateRtpVideoSender(
-      std::map<uint32_t, RtpState> suspended_ssrcs,
+      const std::map<uint32_t, RtpState>& suspended_ssrcs,
       // TODO(holmer): Move states into RtpTransportControllerSend.
       const std::map<uint32_t, RtpPayloadState>& states,
       const RtpConfig& rtp_config,
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index c83d697..b953705 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -347,7 +347,7 @@
 
 RtpVideoSender::RtpVideoSender(
     Clock* clock,
-    std::map<uint32_t, RtpState> suspended_ssrcs,
+    const std::map<uint32_t, RtpState>& suspended_ssrcs,
     const std::map<uint32_t, RtpPayloadState>& states,
     const RtpConfig& rtp_config,
     int rtcp_report_interval_ms,
@@ -371,7 +371,6 @@
           field_trials_.Lookup("WebRTC-GenericCodecDependencyDescriptor"),
           "Enabled")),
       active_(false),
-      suspended_ssrcs_(std::move(suspended_ssrcs)),
       fec_controller_(std::move(fec_controller)),
       fec_allowed_(true),
       rtp_streams_(CreateRtpStreamSenders(clock,
@@ -381,7 +380,7 @@
                                           send_transport,
                                           transport->GetBandwidthObserver(),
                                           transport,
-                                          suspended_ssrcs_,
+                                          suspended_ssrcs,
                                           event_log,
                                           retransmission_limiter,
                                           frame_encryptor,
@@ -421,7 +420,7 @@
     }
   }
 
-  ConfigureSsrcs();
+  ConfigureSsrcs(suspended_ssrcs);
   ConfigureRids();
 
   if (!rtp_config_.mid.empty()) {
@@ -683,7 +682,8 @@
     stream.rtp_rtcp->IncomingRtcpPacket(packet, length);
 }
 
-void RtpVideoSender::ConfigureSsrcs() {
+void RtpVideoSender::ConfigureSsrcs(
+    const std::map<uint32_t, RtpState>& suspended_ssrcs) {
   // Configure regular SSRCs.
   RTC_CHECK(ssrc_to_rtp_module_.empty());
   for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
@@ -691,8 +691,8 @@
     RtpRtcpInterface* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
 
     // Restore RTP state if previous existed.
-    auto it = suspended_ssrcs_.find(ssrc);
-    if (it != suspended_ssrcs_.end())
+    auto it = suspended_ssrcs.find(ssrc);
+    if (it != suspended_ssrcs.end())
       rtp_rtcp->SetRtpState(it->second);
 
     ssrc_to_rtp_module_[ssrc] = rtp_rtcp;
@@ -706,8 +706,8 @@
   for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
     RtpRtcpInterface* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
-    auto it = suspended_ssrcs_.find(ssrc);
-    if (it != suspended_ssrcs_.end())
+    auto it = suspended_ssrcs.find(ssrc);
+    if (it != suspended_ssrcs.end())
       rtp_rtcp->SetRtxState(it->second);
   }
 
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 704ed0e..7e5de98 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -74,7 +74,7 @@
   // Rtp modules are assumed to be sorted in simulcast index order.
   RtpVideoSender(
       Clock* clock,
-      std::map<uint32_t, RtpState> suspended_ssrcs,
+      const std::map<uint32_t, RtpState>& suspended_ssrcs,
       const std::map<uint32_t, RtpPayloadState>& states,
       const RtpConfig& rtp_config,
       int rtcp_report_interval_ms,
@@ -155,7 +155,7 @@
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
   void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
   void ConfigureProtection();
-  void ConfigureSsrcs();
+  void ConfigureSsrcs(const std::map<uint32_t, RtpState>& suspended_ssrcs);
   void ConfigureRids();
   bool NackEnabled() const;
   uint32_t GetPacketizationOverheadRate() const;
@@ -175,8 +175,6 @@
   mutable Mutex mutex_;
   bool active_ RTC_GUARDED_BY(mutex_);
 
-  std::map<uint32_t, RtpState> suspended_ssrcs_;
-
   const std::unique_ptr<FecController> fec_controller_;
   bool fec_allowed_ RTC_GUARDED_BY(mutex_);
 
diff --git a/call/test/mock_rtp_transport_controller_send.h b/call/test/mock_rtp_transport_controller_send.h
index b468aa6..8d637a3 100644
--- a/call/test/mock_rtp_transport_controller_send.h
+++ b/call/test/mock_rtp_transport_controller_send.h
@@ -34,7 +34,7 @@
  public:
   MOCK_METHOD(RtpVideoSenderInterface*,
               CreateRtpVideoSender,
-              ((std::map<uint32_t, RtpState>),
+              ((const std::map<uint32_t, RtpState>&),
                (const std::map<uint32_t, RtpPayloadState>&),
                const RtpConfig&,
                int rtcp_report_interval_ms,