diff --git a/test/BUILD.gn b/test/BUILD.gn
index 773e25f66c..a21637fd7c 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -327,6 +327,7 @@ if (rtc_include_tests) {
 
   rtc_test("test_support_unittests") {
     deps = [
+      ":call_config_utils",
       ":direct_transport",
       ":fake_video_codecs",
       ":fileutils",
@@ -345,6 +346,7 @@ if (rtc_include_tests) {
       "../api/video:builtin_video_bitrate_allocator_factory",
       "../api/video:video_frame",
       "../api/video:video_frame_i420",
+      "../call:video_stream_api",
       "../modules/rtp_rtcp:rtp_rtcp",
       "../modules/video_capture",
       "../modules/video_coding:simulcast_test_fixture_impl",
@@ -358,6 +360,7 @@ if (rtc_include_tests) {
       "//third_party/abseil-cpp/absl/strings",
     ]
     sources = [
+      "call_config_utils_unittest.cc",
       "direct_transport_unittest.cc",
       "fake_vp8_encoder_unittest.cc",
       "frame_generator_unittest.cc",
diff --git a/test/call_config_utils.cc b/test/call_config_utils.cc
index 48d4849a5c..671220b800 100644
--- a/test/call_config_utils.cc
+++ b/test/call_config_utils.cc
@@ -68,5 +68,59 @@ VideoReceiveStream::Config ParseVideoReceiveStreamJsonConfig(
   return receive_config;
 }
 
+Json::Value GenerateVideoReceiveStreamJsonConfig(
+    const VideoReceiveStream::Config& config) {
+  Json::Value root_json;
+
+  root_json["decoders"] = Json::Value(Json::arrayValue);
+  for (const auto& decoder : config.decoders) {
+    Json::Value decoder_json;
+    decoder_json["payload_type"] = decoder.payload_type;
+    decoder_json["payload_name"] = decoder.video_format.name;
+    decoder_json["codec_params"] = Json::Value(Json::arrayValue);
+    for (const auto& codec_param_entry : decoder.video_format.parameters) {
+      Json::Value codec_param_json;
+      codec_param_json[codec_param_entry.first] = codec_param_entry.second;
+      decoder_json["codec_params"].append(codec_param_json);
+    }
+    root_json["decoders"].append(decoder_json);
+  }
+
+  Json::Value rtp_json;
+  rtp_json["remote_ssrc"] = config.rtp.remote_ssrc;
+  rtp_json["local_ssrc"] = config.rtp.local_ssrc;
+  rtp_json["rtcp_mode"] = config.rtp.rtcp_mode == RtcpMode::kCompound
+                              ? "RtcpMode::kCompound"
+                              : "RtcpMode::kReducedSize";
+  rtp_json["remb"] = config.rtp.remb;
+  rtp_json["transport_cc"] = config.rtp.transport_cc;
+  rtp_json["nack"]["rtp_history_ms"] = config.rtp.nack.rtp_history_ms;
+  rtp_json["ulpfec_payload_type"] = config.rtp.ulpfec_payload_type;
+  rtp_json["red_payload_type"] = config.rtp.red_payload_type;
+  rtp_json["rtx_ssrc"] = config.rtp.rtx_ssrc;
+  rtp_json["rtx_payload_types"] = Json::Value(Json::arrayValue);
+
+  for (auto& kv : config.rtp.rtx_associated_payload_types) {
+    Json::Value val;
+    val[std::to_string(kv.first)] = kv.second;
+    rtp_json["rtx_payload_types"].append(val);
+  }
+
+  rtp_json["extensions"] = Json::Value(Json::arrayValue);
+  for (auto& ext : config.rtp.extensions) {
+    Json::Value ext_json;
+    ext_json["uri"] = ext.uri;
+    ext_json["id"] = ext.id;
+    ext_json["encrypt"] = ext.encrypt;
+    rtp_json["extensions"].append(ext_json);
+  }
+  root_json["rtp"] = rtp_json;
+
+  root_json["render_delay_ms"] = config.render_delay_ms;
+  root_json["target_delay_ms"] = config.target_delay_ms;
+
+  return root_json;
+}
+
 }  // namespace test.
 }  // namespace webrtc.
diff --git a/test/call_config_utils.h b/test/call_config_utils.h
index 246c7f810d..186c67fd86 100644
--- a/test/call_config_utils.h
+++ b/test/call_config_utils.h
@@ -23,6 +23,10 @@ VideoReceiveStream::Config ParseVideoReceiveStreamJsonConfig(
     webrtc::Transport* transport,
     const Json::Value& json);
 
+// Serialize a VideoReceiveStream::Config into a Json object.
+Json::Value GenerateVideoReceiveStreamJsonConfig(
+    const VideoReceiveStream::Config& config);
+
 }  // namespace test
 }  // namespace webrtc
 
diff --git a/test/call_config_utils_unittest.cc b/test/call_config_utils_unittest.cc
new file mode 100644
index 0000000000..afc48c4c9b
--- /dev/null
+++ b/test/call_config_utils_unittest.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/call_config_utils.h"
+
+#include "call/video_receive_stream.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
+  VideoReceiveStream::Config recv_config(nullptr);
+
+  VideoReceiveStream::Decoder decoder;
+  decoder.payload_type = 10;
+  decoder.video_format.name = "test";
+  decoder.video_format.parameters["99"] = 98;
+  recv_config.decoders.push_back(decoder);
+  recv_config.render_delay_ms = 10;
+  recv_config.target_delay_ms = 15;
+  recv_config.rtp.remote_ssrc = 100;
+  recv_config.rtp.local_ssrc = 101;
+  recv_config.rtp.rtcp_mode = RtcpMode::kCompound;
+  recv_config.rtp.remb = false;
+  recv_config.rtp.transport_cc = false;
+  recv_config.rtp.nack.rtp_history_ms = 150;
+  recv_config.rtp.red_payload_type = 50;
+  recv_config.rtp.rtx_ssrc = 1000;
+  recv_config.rtp.rtx_associated_payload_types[10] = 10;
+  recv_config.rtp.extensions.emplace_back("uri", 128, true);
+
+  VideoReceiveStream::Config unmarshaled_config =
+      ParseVideoReceiveStreamJsonConfig(
+          nullptr, GenerateVideoReceiveStreamJsonConfig(recv_config));
+
+  EXPECT_EQ(recv_config.decoders[0].payload_type,
+            unmarshaled_config.decoders[0].payload_type);
+  EXPECT_EQ(recv_config.decoders[0].video_format.name,
+            unmarshaled_config.decoders[0].video_format.name);
+  EXPECT_EQ(recv_config.decoders[0].video_format.parameters,
+            unmarshaled_config.decoders[0].video_format.parameters);
+  EXPECT_EQ(recv_config.render_delay_ms, unmarshaled_config.render_delay_ms);
+  EXPECT_EQ(recv_config.target_delay_ms, unmarshaled_config.target_delay_ms);
+  EXPECT_EQ(recv_config.rtp.remote_ssrc, unmarshaled_config.rtp.remote_ssrc);
+  EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc);
+  EXPECT_EQ(recv_config.rtp.rtcp_mode, unmarshaled_config.rtp.rtcp_mode);
+  EXPECT_EQ(recv_config.rtp.remb, unmarshaled_config.rtp.remb);
+  EXPECT_EQ(recv_config.rtp.transport_cc, unmarshaled_config.rtp.transport_cc);
+  EXPECT_EQ(recv_config.rtp.nack.rtp_history_ms,
+            unmarshaled_config.rtp.nack.rtp_history_ms);
+  EXPECT_EQ(recv_config.rtp.red_payload_type,
+            unmarshaled_config.rtp.red_payload_type);
+  EXPECT_EQ(recv_config.rtp.rtx_ssrc, unmarshaled_config.rtp.rtx_ssrc);
+  EXPECT_EQ(recv_config.rtp.rtx_associated_payload_types,
+            unmarshaled_config.rtp.rtx_associated_payload_types);
+  EXPECT_EQ(recv_config.rtp.extensions, recv_config.rtp.extensions);
+}
+
+}  // namespace test
+}  // namespace webrtc