diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 928b6155c7..99dfa54bc3 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -35,6 +35,9 @@ if (is_ios) {
       #"objc/RTCAudioTrack+Private.h",
       #"objc/RTCAudioTrack.h",
       #"objc/RTCAudioTrack.mm",
+      #"objc/RTCConfiguration+Private.h",
+      #"objc/RTCConfiguration.h",
+      #"objc/RTCConfiguration.mm",
       #"objc/RTCDataChannel+Private.h",
       #"objc/RTCDataChannel.h",
       #"objc/RTCDataChannel.mm",
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index ea31c17cb3..3e2bf41263 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -25,6 +25,9 @@
             'objc/RTCAudioTrack+Private.h',
             'objc/RTCAudioTrack.h',
             'objc/RTCAudioTrack.mm',
+            'objc/RTCConfiguration+Private.h',
+            'objc/RTCConfiguration.h',
+            'objc/RTCConfiguration.mm',
             'objc/RTCDataChannel+Private.h',
             'objc/RTCDataChannel.h',
             'objc/RTCDataChannel.mm',
diff --git a/webrtc/api/api_tests.gyp b/webrtc/api/api_tests.gyp
index c2c18bc693..f073cea454 100644
--- a/webrtc/api/api_tests.gyp
+++ b/webrtc/api/api_tests.gyp
@@ -19,6 +19,7 @@
             '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
           ],
           'sources': [
+            'objctests/RTCConfigurationTest.mm',
             'objctests/RTCIceCandidateTest.mm',
             'objctests/RTCIceServerTest.mm',
             'objctests/RTCMediaConstraintsTest.mm',
diff --git a/webrtc/api/objc/RTCConfiguration+Private.h b/webrtc/api/objc/RTCConfiguration+Private.h
new file mode 100644
index 0000000000..5936ee26e0
--- /dev/null
+++ b/webrtc/api/objc/RTCConfiguration+Private.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2015 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.
+ */
+
+#import "RTCConfiguration.h"
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCConfiguration ()
+
+/**
+ * RTCConfiguration struct representation of this RTCConfiguration. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+    webrtc::PeerConnectionInterface::RTCConfiguration nativeConfiguration;
+
+- (instancetype)initWithNativeConfiguration:
+    (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfiguration;
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)
+    nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy;
+
++ (RTCIceTransportPolicy)transportPolicyForTransportsType:
+    (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
+
++ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
+    (RTCBundlePolicy)policy;
+
++ (RTCBundlePolicy)bundlePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
+
++ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
+    (RTCRtcpMuxPolicy)policy;
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
+
++ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
+    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy;
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
+
++ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCConfiguration.h b/webrtc/api/objc/RTCConfiguration.h
new file mode 100644
index 0000000000..f97daf6d8e
--- /dev/null
+++ b/webrtc/api/objc/RTCConfiguration.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright 2015 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.
+ */
+
+#import <Foundation/Foundation.h>
+
+@class RTCIceServer;
+
+/**
+ * Represents the ice transport policy. This exposes the same states in C++,
+ * which include one more state than what exists in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
+  RTCIceTransportPolicyNone,
+  RTCIceTransportPolicyRelay,
+  RTCIceTransportPolicyNoHost,
+  RTCIceTransportPolicyAll
+};
+
+/** Represents the bundle policy. */
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  RTCBundlePolicyBalanced,
+  RTCBundlePolicyMaxCompat,
+  RTCBundlePolicyMaxBundle
+};
+
+/** Represents the rtcp mux policy. */
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
+  RTCRtcpMuxPolicyNegotiate,
+  RTCRtcpMuxPolicyRequire
+};
+
+/** Represents the tcp candidate policy. */
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  RTCTcpCandidatePolicyEnabled,
+  RTCTcpCandidatePolicyDisabled
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCConfiguration : NSObject
+
+/** An array of Ice Servers available to be used by ICE. */
+@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
+
+/** Which candidates the ICE agent is allowed to use. The W3C calls it
+ * |iceTransportPolicy|, while in C++ it is called |type|. */
+@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
+
+/** The media-bundling policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+
+/** The rtcp-mux policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithIceServers:
+    (nullable NSArray<RTCIceServer *> *)iceServers
+                iceTransportPolicy:(RTCIceTransportPolicy)iceTransportPolicy
+                      bundlePolicy:(RTCBundlePolicy)bundlePolicy
+                     rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
+                tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
+       audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
+     iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
+iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCConfiguration.mm b/webrtc/api/objc/RTCConfiguration.mm
new file mode 100644
index 0000000000..ec59ca2d53
--- /dev/null
+++ b/webrtc/api/objc/RTCConfiguration.mm
@@ -0,0 +1,276 @@
+/*
+ *  Copyright 2015 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.
+ */
+
+#import "RTCConfiguration.h"
+
+#import "webrtc/api/objc/RTCConfiguration+Private.h"
+#import "webrtc/api/objc/RTCIceServer+Private.h"
+
+@implementation RTCConfiguration
+
+@synthesize iceServers = _iceServers;
+@synthesize iceTransportPolicy = _iceTransportPolicy;
+@synthesize bundlePolicy = _bundlePolicy;
+@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
+@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
+@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
+@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
+@synthesize iceBackupCandidatePairPingInterval =
+    _iceBackupCandidatePairPingInterval;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _iceServers = [NSMutableArray array];
+    // Copy defaults.
+    webrtc::PeerConnectionInterface::RTCConfiguration config;
+    _iceTransportPolicy =
+        [[self class] transportPolicyForTransportsType:config.type];
+    _bundlePolicy =
+        [[self class] bundlePolicyForNativePolicy:config.bundle_policy];
+    _rtcpMuxPolicy =
+        [[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
+    _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
+        config.tcp_candidate_policy];
+    _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
+    _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
+    _iceBackupCandidatePairPingInterval =
+        config.ice_backup_candidate_pair_ping_interval;
+  }
+  return self;
+}
+
+- (instancetype)initWithIceServers:(NSArray<RTCIceServer *> *)iceServers
+                    iceTransportPolicy:(RTCIceTransportPolicy)iceTransportPolicy
+                          bundlePolicy:(RTCBundlePolicy)bundlePolicy
+                         rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
+                    tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
+           audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
+         iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
+    iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval {
+  if (self = [self init]) {
+    if (iceServers) {
+      _iceServers = [iceServers copy];
+    }
+    _iceTransportPolicy = iceTransportPolicy;
+    _bundlePolicy = bundlePolicy;
+    _rtcpMuxPolicy = rtcpMuxPolicy;
+    _tcpCandidatePolicy = tcpCandidatePolicy;
+    _audioJitterBufferMaxPackets = audioJitterBufferMaxPackets;
+    _iceConnectionReceivingTimeout = iceConnectionReceivingTimeout;
+    _iceBackupCandidatePairPingInterval = iceBackupCandidatePairPingInterval;
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:
+      @"RTCConfiguration: {\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n}\n",
+      _iceServers,
+      [[self class] stringForTransportPolicy:_iceTransportPolicy],
+      [[self class] stringForBundlePolicy:_bundlePolicy],
+      [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
+      [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
+      _audioJitterBufferMaxPackets,
+      _iceConnectionReceivingTimeout,
+      _iceBackupCandidatePairPingInterval];
+}
+
+#pragma mark - Private
+
+- (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfiguration {
+  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
+
+  for (RTCIceServer *iceServer in _iceServers) {
+    nativeConfig.servers.push_back(iceServer.iceServer);
+  }
+  nativeConfig.type =
+      [[self class] nativeTransportsTypeForTransportPolicy:_iceTransportPolicy];
+  nativeConfig.bundle_policy =
+      [[self class] nativeBundlePolicyForPolicy:_bundlePolicy];
+  nativeConfig.rtcp_mux_policy =
+      [[self class] nativeRtcpMuxPolicyForPolicy:_rtcpMuxPolicy];
+  nativeConfig.tcp_candidate_policy =
+      [[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
+  nativeConfig.audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
+  nativeConfig.ice_connection_receiving_timeout =
+      _iceConnectionReceivingTimeout;
+  nativeConfig.ice_backup_candidate_pair_ping_interval =
+      _iceBackupCandidatePairPingInterval;
+
+  return nativeConfig;
+}
+
+- (instancetype)initWithNativeConfiguration:
+    (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfig {
+  NSMutableArray *iceServers =
+        [NSMutableArray arrayWithCapacity:nativeConfig.servers.size()];
+  for (auto const &server : nativeConfig.servers) {
+    RTCIceServer *iceServer =
+        [[RTCIceServer alloc] initWithNativeServer:server];
+    [iceServers addObject:iceServer];
+  }
+
+  if (self = [self init]) {
+    if (iceServers) {
+      _iceServers = [iceServers copy];
+    }
+    _iceTransportPolicy =
+        [[self class] transportPolicyForTransportsType:nativeConfig.type];
+    _bundlePolicy =
+        [[self class] bundlePolicyForNativePolicy:nativeConfig.bundle_policy];
+    _rtcpMuxPolicy = [[self class] rtcpMuxPolicyForNativePolicy:
+        nativeConfig.rtcp_mux_policy];
+    _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
+        nativeConfig.tcp_candidate_policy];
+    _audioJitterBufferMaxPackets = nativeConfig.audio_jitter_buffer_max_packets;
+    _iceConnectionReceivingTimeout =
+        nativeConfig.ice_connection_receiving_timeout;
+    _iceBackupCandidatePairPingInterval =
+        nativeConfig.ice_backup_candidate_pair_ping_interval;
+  }
+
+  return self;
+}
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)
+    nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy {
+  switch (policy) {
+    case RTCIceTransportPolicyNone:
+      return webrtc::PeerConnectionInterface::kNone;
+    case RTCIceTransportPolicyRelay:
+      return webrtc::PeerConnectionInterface::kRelay;
+    case RTCIceTransportPolicyNoHost:
+      return webrtc::PeerConnectionInterface::kNoHost;
+    case RTCIceTransportPolicyAll:
+      return webrtc::PeerConnectionInterface::kAll;
+  }
+}
+
++ (RTCIceTransportPolicy)transportPolicyForTransportsType:
+    (webrtc::PeerConnectionInterface::IceTransportsType)nativeType {
+  switch (nativeType) {
+    case webrtc::PeerConnectionInterface::kNone:
+      return RTCIceTransportPolicyNone;
+    case webrtc::PeerConnectionInterface::kRelay:
+      return RTCIceTransportPolicyRelay;
+    case webrtc::PeerConnectionInterface::kNoHost:
+      return RTCIceTransportPolicyNoHost;
+    case webrtc::PeerConnectionInterface::kAll:
+      return RTCIceTransportPolicyAll;
+  }
+}
+
++ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy {
+  switch (policy) {
+    case RTCIceTransportPolicyNone:
+      return @"NONE";
+    case RTCIceTransportPolicyRelay:
+      return @"RELAY";
+    case RTCIceTransportPolicyNoHost:
+      return @"NO_HOST";
+    case RTCIceTransportPolicyAll:
+      return @"ALL";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
+    (RTCBundlePolicy)policy {
+  switch (policy) {
+    case RTCBundlePolicyBalanced:
+      return webrtc::PeerConnectionInterface::kBundlePolicyBalanced;
+    case RTCBundlePolicyMaxCompat:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
+    case RTCBundlePolicyMaxBundle:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
+  }
+}
+
++ (RTCBundlePolicy)bundlePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kBundlePolicyBalanced:
+      return RTCBundlePolicyBalanced;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat:
+      return RTCBundlePolicyMaxCompat;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle:
+      return RTCBundlePolicyMaxBundle;
+  }
+}
+
++ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy {
+  switch (policy) {
+    case RTCBundlePolicyBalanced:
+      return @"BALANCED";
+    case RTCBundlePolicyMaxCompat:
+      return @"MAX_COMPAT";
+    case RTCBundlePolicyMaxBundle:
+      return @"MAX_BUNDLE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
+    (RTCRtcpMuxPolicy)policy {
+  switch (policy) {
+    case RTCRtcpMuxPolicyNegotiate:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
+    case RTCRtcpMuxPolicyRequire:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  }
+}
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate:
+      return RTCRtcpMuxPolicyNegotiate;
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire:
+      return RTCRtcpMuxPolicyRequire;
+  }
+}
+
++ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy {
+  switch (policy) {
+    case RTCRtcpMuxPolicyNegotiate:
+      return @"NEGOTIATE";
+    case RTCRtcpMuxPolicyRequire:
+      return @"REQUIRE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
+    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy {
+  switch (policy) {
+    case RTCTcpCandidatePolicyEnabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
+    case RTCTcpCandidatePolicyDisabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled;
+  }
+}
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled:
+      return RTCTcpCandidatePolicyEnabled;
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled:
+      return RTCTcpCandidatePolicyDisabled;
+  }
+}
+
++ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy {
+  switch (policy) {
+    case RTCTcpCandidatePolicyEnabled:
+      return @"TCP_ENABLED";
+    case RTCTcpCandidatePolicyDisabled:
+      return @"TCP_DISABLED";
+  }
+}
+
+@end
diff --git a/webrtc/api/objc/RTCIceServer+Private.h b/webrtc/api/objc/RTCIceServer+Private.h
index 59f5a92dff..3890567042 100644
--- a/webrtc/api/objc/RTCIceServer+Private.h
+++ b/webrtc/api/objc/RTCIceServer+Private.h
@@ -23,6 +23,10 @@ NS_ASSUME_NONNULL_BEGIN
 @property(nonatomic, readonly)
     webrtc::PeerConnectionInterface::IceServer iceServer;
 
+/** Initialize an RTCIceServer from a native IceServer. */
+- (instancetype)initWithNativeServer:
+    (webrtc::PeerConnectionInterface::IceServer)nativeServer;
+
 @end
 
 NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCIceServer.mm b/webrtc/api/objc/RTCIceServer.mm
index 7a898e06d5..057c69603b 100644
--- a/webrtc/api/objc/RTCIceServer.mm
+++ b/webrtc/api/objc/RTCIceServer.mm
@@ -61,4 +61,19 @@
   return iceServer;
 }
 
+- (instancetype)initWithNativeServer:
+    (webrtc::PeerConnectionInterface::IceServer)nativeServer {
+  NSMutableArray *urls =
+      [NSMutableArray arrayWithCapacity:nativeServer.urls.size()];
+  for (auto const &url : nativeServer.urls) {
+    [urls addObject:[NSString stringForStdString:url]];
+  }
+  NSString *username = [NSString stringForStdString:nativeServer.username];
+  NSString *credential = [NSString stringForStdString:nativeServer.password];
+  self = [self initWithURLStrings:urls
+                         username:username
+                       credential:credential];
+  return self;
+}
+
 @end
diff --git a/webrtc/api/objctests/RTCConfigurationTest.mm b/webrtc/api/objctests/RTCConfigurationTest.mm
new file mode 100644
index 0000000000..429ce11fec
--- /dev/null
+++ b/webrtc/api/objctests/RTCConfigurationTest.mm
@@ -0,0 +1,118 @@
+/*
+ *  Copyright 2015 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.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include <vector>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCConfiguration.h"
+#import "webrtc/api/objc/RTCConfiguration+Private.h"
+#import "webrtc/api/objc/RTCIceServer.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCConfigurationTest : NSObject
+- (void)testConversionToNativeConfiguration;
+- (void)testInitFromNativeConfiguration;
+@end
+
+@implementation RTCConfigurationTest
+
+- (void)testConversionToNativeConfiguration {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
+
+  RTCConfiguration *config =
+      [[RTCConfiguration alloc] initWithIceServers:@[ server ]
+                                iceTransportPolicy:RTCIceTransportPolicyRelay
+                                      bundlePolicy:RTCBundlePolicyMaxBundle
+                                     rtcpMuxPolicy:RTCRtcpMuxPolicyNegotiate
+                                tcpCandidatePolicy:RTCTcpCandidatePolicyDisabled
+                       audioJitterBufferMaxPackets:60
+                     iceConnectionReceivingTimeout:1
+                iceBackupCandidatePairPingInterval:2];
+
+  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig =
+      config.nativeConfiguration;
+  EXPECT_EQ(1u, nativeConfig.servers.size());
+  webrtc::PeerConnectionInterface::IceServer nativeServer =
+      nativeConfig.servers.front();
+  EXPECT_EQ(1u, nativeServer.urls.size());
+  EXPECT_EQ("stun:stun1.example.net", nativeServer.urls.front());
+
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig.type);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle,
+            nativeConfig.bundle_policy);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate,
+            nativeConfig.rtcp_mux_policy);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled,
+            nativeConfig.tcp_candidate_policy);
+  EXPECT_EQ(60, nativeConfig.audio_jitter_buffer_max_packets);
+  EXPECT_EQ(1, nativeConfig.ice_connection_receiving_timeout);
+  EXPECT_EQ(2, nativeConfig.ice_backup_candidate_pair_ping_interval);
+}
+
+- (void)testInitFromNativeConfiguration {
+  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
+
+  webrtc::PeerConnectionInterface::IceServer nativeServer;
+  nativeServer.username = "username";
+  nativeServer.password = "password";
+  nativeServer.urls.push_back("stun:stun.example.net");
+  webrtc::PeerConnectionInterface::IceServers servers { nativeServer };
+
+  nativeConfig.servers = servers;
+  nativeConfig.type = webrtc::PeerConnectionInterface::kNoHost;
+  nativeConfig.bundle_policy =
+      webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
+  nativeConfig.rtcp_mux_policy =
+      webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  nativeConfig.tcp_candidate_policy =
+      webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
+  nativeConfig.audio_jitter_buffer_max_packets = 40;
+  nativeConfig.ice_connection_receiving_timeout =
+      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined;
+  nativeConfig.ice_backup_candidate_pair_ping_interval =
+      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined;
+
+  RTCConfiguration *config =
+      [[RTCConfiguration alloc] initWithNativeConfiguration:nativeConfig];
+
+  EXPECT_EQ(1u, config.iceServers.count);
+  RTCIceServer *server = config.iceServers.firstObject;
+  EXPECT_EQ(1u, server.urlStrings.count);
+  EXPECT_TRUE([@"stun:stun.example.net" isEqualToString:
+      server.urlStrings.firstObject]);
+
+  EXPECT_EQ(RTCIceTransportPolicyNoHost, config.iceTransportPolicy);
+  EXPECT_EQ(RTCBundlePolicyMaxCompat, config.bundlePolicy);
+  EXPECT_EQ(RTCRtcpMuxPolicyRequire, config.rtcpMuxPolicy);
+  EXPECT_EQ(RTCTcpCandidatePolicyEnabled, config.tcpCandidatePolicy);
+  EXPECT_EQ(40, config.audioJitterBufferMaxPackets);
+  EXPECT_EQ(-1, config.iceConnectionReceivingTimeout);
+  EXPECT_EQ(-1, config.iceBackupCandidatePairPingInterval);
+}
+
+@end
+
+TEST(RTCConfigurationTest, NativeConfigurationConversionTest) {
+  @autoreleasepool {
+    RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
+    [test testConversionToNativeConfiguration];
+  }
+}
+
+TEST(RTCConfigurationTest, InitFromConfigurationTest) {
+  @autoreleasepool {
+    RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
+    [test testInitFromNativeConfiguration];
+  }
+}
diff --git a/webrtc/api/objctests/RTCIceServerTest.mm b/webrtc/api/objctests/RTCIceServerTest.mm
index 5fa43f8447..2e6fb25486 100644
--- a/webrtc/api/objctests/RTCIceServerTest.mm
+++ b/webrtc/api/objctests/RTCIceServerTest.mm
@@ -16,11 +16,13 @@
 
 #import "webrtc/api/objc/RTCIceServer.h"
 #import "webrtc/api/objc/RTCIceServer+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
 
 @interface RTCIceServerTest : NSObject
 - (void)testOneURLServer;
 - (void)testTwoURLServer;
 - (void)testPasswordCredential;
+- (void)testInitFromNativeServer;
 @end
 
 @implementation RTCIceServerTest
@@ -30,7 +32,7 @@
       @"stun:stun1.example.net" ]];
 
   webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
-  EXPECT_EQ((size_t)1, iceStruct.urls.size());
+  EXPECT_EQ(1u, iceStruct.urls.size());
   EXPECT_EQ("stun:stun1.example.net", iceStruct.urls.front());
   EXPECT_EQ("", iceStruct.username);
   EXPECT_EQ("", iceStruct.password);
@@ -41,7 +43,7 @@
       @"turn1:turn1.example.net", @"turn2:turn2.example.net" ]];
 
   webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
-  EXPECT_EQ((size_t)2, iceStruct.urls.size());
+  EXPECT_EQ(2u, iceStruct.urls.size());
   EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
   EXPECT_EQ("turn2:turn2.example.net", iceStruct.urls.back());
   EXPECT_EQ("", iceStruct.username);
@@ -54,12 +56,27 @@
                 username:@"username"
               credential:@"credential"];
   webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
-  EXPECT_EQ((size_t)1, iceStruct.urls.size());
+  EXPECT_EQ(1u, iceStruct.urls.size());
   EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
   EXPECT_EQ("username", iceStruct.username);
   EXPECT_EQ("credential", iceStruct.password);
 }
 
+- (void)testInitFromNativeServer {
+  webrtc::PeerConnectionInterface::IceServer nativeServer;
+  nativeServer.username = "username";
+  nativeServer.password = "password";
+  nativeServer.urls.push_back("stun:stun.example.net");
+
+  RTCIceServer *iceServer =
+      [[RTCIceServer alloc] initWithNativeServer:nativeServer];
+  EXPECT_EQ(1u, iceServer.urlStrings.count);
+  EXPECT_EQ("stun:stun.example.net",
+      [NSString stdStringForString:iceServer.urlStrings.firstObject]);
+  EXPECT_EQ("username", [NSString stdStringForString:iceServer.username]);
+  EXPECT_EQ("password", [NSString stdStringForString:iceServer.credential]);
+}
+
 @end
 
 TEST(RTCIceServerTest, OneURLTest) {
@@ -82,3 +99,10 @@ TEST(RTCIceServerTest, PasswordCredentialTest) {
     [test testPasswordCredential];
   }
 }
+
+TEST(RTCIceServerTest, InitFromNativeServerTest) {
+  @autoreleasepool {
+    RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+    [test testInitFromNativeServer];
+  }
+}