diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index b44bbe54ac..654e810dad 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -65,6 +65,7 @@ rtc_static_library("audio") {
     "../common_audio:common_audio_c",
     "../logging:rtc_event_audio",
     "../logging:rtc_event_log_api",
+    "../logging:rtc_stream_config",
     "../modules/audio_coding",
     "../modules/audio_coding:audio_format_conversion",
     "../modules/audio_coding:audio_network_adaptor_config",
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index f778745e31..f11dace832 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -28,6 +28,9 @@
 #include "call/rtp_transport_controller_send_interface.h"
 #include "common_audio/vad/include/vad.h"
 #include "common_types.h"  // NOLINT(build/include)
+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/rtc_stream_config.h"
 #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "rtc_base/checks.h"
@@ -69,6 +72,39 @@ std::unique_ptr<voe::ChannelSendProxy> CreateChannelAndProxy(
           worker_queue, module_process_thread, media_transport, rtcp_rtt_stats,
           event_log, frame_encryptor, crypto_options, extmap_allow_mixed));
 }
+
+void UpdateEventLogStreamConfig(RtcEventLog* event_log,
+                                const AudioSendStream::Config& config,
+                                const AudioSendStream::Config* old_config) {
+  using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
+  // Only update if any of the things we log have changed.
+  auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
+                                const absl::optional<SendCodecSpec>& b) {
+    if (a.has_value() && b.has_value()) {
+      return a->format.name == b->format.name &&
+             a->payload_type == b->payload_type;
+    }
+    return !a.has_value() && !b.has_value();
+  };
+
+  if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
+      config.rtp.extensions == old_config->rtp.extensions &&
+      payload_types_equal(config.send_codec_spec,
+                          old_config->send_codec_spec)) {
+    return;
+  }
+
+  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
+  rtclog_config->local_ssrc = config.rtp.ssrc;
+  rtclog_config->rtp_extensions = config.rtp.extensions;
+  if (config.send_codec_spec) {
+    rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
+                                       config.send_codec_spec->payload_type, 0);
+  }
+  event_log->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
+      std::move(rtclog_config)));
+}
+
 }  // namespace
 
 // Helper class to track the actively sending lifetime of this stream.
@@ -221,6 +257,9 @@ void AudioSendStream::ConfigureStream(
     bool first_time) {
   RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
                    << new_config.ToString();
+  UpdateEventLogStreamConfig(stream->event_log_, new_config,
+                             first_time ? nullptr : &stream->config_);
+
   const auto& channel_proxy = stream->channel_proxy_;
   const auto& old_config = stream->config_;
 
diff --git a/call/call.cc b/call/call.cc
index 1ecaa613e9..1233ecdb7c 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -30,7 +30,6 @@
 #include "call/rtp_stream_receiver_controller.h"
 #include "call/rtp_transport_controller_send.h"
 #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
-#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
 #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
 #include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
 #include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
@@ -147,18 +146,6 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
   return rtclog_config;
 }
 
-std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
-    const AudioSendStream::Config& config) {
-  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
-  rtclog_config->local_ssrc = config.rtp.ssrc;
-  rtclog_config->rtp_extensions = config.rtp.extensions;
-  if (config.send_codec_spec) {
-    rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
-                                       config.send_codec_spec->payload_type, 0);
-  }
-  return rtclog_config;
-}
-
 }  // namespace
 
 namespace internal {
@@ -579,9 +566,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
-  event_log_->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
-      CreateRtcLogStreamConfig(config)));
-
+  // Stream config is logged in AudioSendStream::ConfigureStream, as it may
+  // change during the stream's lifetime.
   absl::optional<RtpState> suspended_rtp_state;
   {
     const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);