diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index fd320324b8..0929fd9675 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -842,15 +842,16 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer,
                                const RTPHeader& header,
                                bool is_rtx,
                                bool is_retransmit) {
-  CriticalSectionScoped lock(statistics_crit_.get());
   StreamDataCounters* counters;
-  uint32_t ssrc;
+  // Get ssrc before taking statistics_crit_ to avoid possible deadlock.
+  uint32_t ssrc = SSRC();
+
+  CriticalSectionScoped lock(statistics_crit_.get());
   if (is_rtx) {
     counters = &rtx_rtp_stats_;
     ssrc = ssrc_rtx_;
   } else {
     counters = &rtp_stats_;
-    ssrc = ssrc_;
   }
 
   bitrate_sent_.Update(size);