diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index a263a73721..7e2093de0e 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -267,8 +267,12 @@ NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
       prev_lifetime_stats_ = lifetime_stats;
       const bool no_more_packets_to_decode =
           !input_->NextPacketTime() && !operations_state.next_packet_available;
-      result.is_simulation_finished =
-          no_more_packets_to_decode || input_->ended();
+      // End the simulation if the gap is too large. This indicates an issue
+      // with the event log file.
+      const bool simulation_step_too_large = result.simulation_step_ms > 1000;
+      result.is_simulation_finished = simulation_step_too_large ||
+                                      no_more_packets_to_decode ||
+                                      input_->ended();
       prev_ops_state_ = operations_state;
       return result;
     }
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index 789cc52f76..4719922b98 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -138,6 +138,11 @@ bool RtcEventLogSource::Initialize(const ParsedRtcEventLog& parsed_log,
       continue;
     }
     event_processor.AddEvents(rtp_packets.incoming_packets, handle_rtp_packet);
+    // If no SSRC filter has been set, use the first SSRC only. The simulator
+    // does not work properly with interleaved packets from multiple SSRCs.
+    if (!ssrc_filter.has_value()) {
+      ssrc_filter = rtp_packets.ssrc;
+    }
   }
 
   for (const auto& audio_playouts : parsed_log.audio_playout_events()) {