diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 078aea66a9..7f0d825610 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -246,6 +246,12 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt,
                                            int64_t probing_interval_ms) {
+  // A send stream may be allocated a bitrate of zero if the allocator decides
+  // to disable it. For now we ignore this decision and keep sending on min
+  // bitrate.
+  if (bitrate_bps == 0) {
+    bitrate_bps = config_.min_bitrate_bps;
+  }
   RTC_DCHECK_GE(bitrate_bps,
                 static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5fcacd16d0..e381183b12 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -374,7 +374,7 @@ Call::Call(const Call::Config& config,
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
-  RTC_DCHECK_GT(config.bitrate_config.start_bitrate_bps,
+  RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,