diff --git a/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index f99a9daa6d..9a6ca47dba 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/talk/app/webrtc/webrtcsdp.cc
@@ -668,8 +668,9 @@ static int GetCandidatePreferenceFromType(const std::string& type) {
   return preference;
 }
 
-// Get ip and port of the default destination from the |candidates| with
-// the given value of |component_id|.
+// Get ip and port of the default destination from the |candidates| with the
+// given value of |component_id|. The default candidate should be the one most
+// likely to work, typically IPv4 relay.
 // RFC 5245
 // The value of |component_id| currently supported are 1 (RTP) and 2 (RTCP).
 // TODO: Decide the default destination in webrtcsession and
@@ -682,25 +683,35 @@ static void GetDefaultDestination(
   *port = kDummyPort;
   *ip = kDummyAddress;
   int current_preference = kPreferenceUnknown;
+  int current_family = AF_UNSPEC;
   for (std::vector<Candidate>::const_iterator it = candidates.begin();
        it != candidates.end(); ++it) {
     if (it->component() != component_id) {
       continue;
     }
-    const int preference = GetCandidatePreferenceFromType(it->type());
-    // See if this candidate is more preferable then the current one.
-    if (preference <= current_preference) {
+    // Default destination should be UDP only.
+    if (it->protocol() != cricket::UDP_PROTOCOL_NAME) {
+      continue;
+    }
+    const int preference = GetCandidatePreferenceFromType(it->type());
+    const int family = it->address().ipaddr().family();
+    // See if this candidate is more preferable then the current one if it's the
+    // same family. Or if the current family is IPv4 already so we could safely
+    // ignore all IPv6 ones. WebRTC bug 4269.
+    // http://code.google.com/p/webrtc/issues/detail?id=4269
+    if ((preference <= current_preference && current_family == family) ||
+        (current_family == AF_INET && family == AF_INET6)) {
       continue;
     }
-    current_preference = preference;
-    *port = it->address().PortAsString();
-    *ip = it->address().ipaddr().ToString();
-    int family = it->address().ipaddr().family();
     if (family == AF_INET) {
       addr_type->assign(kConnectionIpv4Addrtype);
     } else if (family == AF_INET6) {
       addr_type->assign(kConnectionIpv6Addrtype);
     }
+    current_preference = preference;
+    current_family = family;
+    *port = it->address().PortAsString();
+    *ip = it->address().ipaddr().ToString();
   }
 }
 
diff --git a/talk/app/webrtc/webrtcsdp_unittest.cc b/talk/app/webrtc/webrtcsdp_unittest.cc
index 4a2aab1128..e62294416a 100644
--- a/talk/app/webrtc/webrtcsdp_unittest.cc
+++ b/talk/app/webrtc/webrtcsdp_unittest.cc
@@ -1363,6 +1363,127 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionEmpty) {
   EXPECT_EQ("", webrtc::SdpSerialize(jdesc_empty));
 }
 
+// This tests serialization of SDP with only IPv6 candidates and verifies that
+// IPv6 is used as default address in c line according to preference.
+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithIPv6Only) {
+  // Only test 1 m line.
+  desc_.RemoveContentByName("video_content_name");
+  // Stun has a high preference than local host.
+  cricket::Candidate candidate1(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp",
+      rtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
+      cricket::STUN_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  cricket::Candidate candidate2(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp",
+      rtc::SocketAddress("::2", 1235), kCandidatePriority, "", "",
+      cricket::LOCAL_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  JsepSessionDescription jdesc(kDummyString);
+  ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+  // Only add the candidates to audio m line.
+  JsepIceCandidate jice1("audio_content_name", 0, candidate1);
+  JsepIceCandidate jice2("audio_content_name", 0, candidate2);
+  ASSERT_TRUE(jdesc.AddCandidate(&jice1));
+  ASSERT_TRUE(jdesc.AddCandidate(&jice2));
+  std::string message = webrtc::SdpSerialize(jdesc);
+
+  // Audio line should have a c line like this one.
+  EXPECT_NE(message.find("c=IN IP6 ::1"), std::string::npos);
+  // Shouldn't have a IP4 c line.
+  EXPECT_EQ(message.find("c=IN IP4"), std::string::npos);
+}
+
+// This tests serialization of SDP with both IPv4 and IPv6 candidates and
+// verifies that IPv4 is used as default address in c line even if the
+// preference of IPv4 is lower.
+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBothIPFamilies) {
+  // Only test 1 m line.
+  desc_.RemoveContentByName("video_content_name");
+  cricket::Candidate candidate_v4(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp",
+      rtc::SocketAddress("192.168.1.5", 1234), kCandidatePriority, "", "",
+      cricket::STUN_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  cricket::Candidate candidate_v6(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp",
+      rtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
+      cricket::LOCAL_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  JsepSessionDescription jdesc(kDummyString);
+  ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+  // Only add the candidates to audio m line.
+  JsepIceCandidate jice_v4("audio_content_name", 0, candidate_v4);
+  JsepIceCandidate jice_v6("audio_content_name", 0, candidate_v6);
+  ASSERT_TRUE(jdesc.AddCandidate(&jice_v4));
+  ASSERT_TRUE(jdesc.AddCandidate(&jice_v6));
+  std::string message = webrtc::SdpSerialize(jdesc);
+
+  // Audio line should have a c line like this one.
+  EXPECT_NE(message.find("c=IN IP4 192.168.1.5"), std::string::npos);
+  // Shouldn't have a IP6 c line.
+  EXPECT_EQ(message.find("c=IN IP6"), std::string::npos);
+}
+
+// This tests serialization of SDP with both UDP and TCP candidates and
+// verifies that UDP is used as default address in c line even if the
+// preference of UDP is lower.
+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBothProtocols) {
+  // Only test 1 m line.
+  desc_.RemoveContentByName("video_content_name");
+  // Stun has a high preference than local host.
+  cricket::Candidate candidate1(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+      rtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
+      cricket::STUN_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  cricket::Candidate candidate2(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp",
+      rtc::SocketAddress("fe80::1234:5678:abcd:ef12", 1235), kCandidatePriority,
+      "", "", cricket::LOCAL_PORT_TYPE, kCandidateGeneration,
+      kCandidateFoundation1);
+  JsepSessionDescription jdesc(kDummyString);
+  ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+  // Only add the candidates to audio m line.
+  JsepIceCandidate jice1("audio_content_name", 0, candidate1);
+  JsepIceCandidate jice2("audio_content_name", 0, candidate2);
+  ASSERT_TRUE(jdesc.AddCandidate(&jice1));
+  ASSERT_TRUE(jdesc.AddCandidate(&jice2));
+  std::string message = webrtc::SdpSerialize(jdesc);
+
+  // Audio line should have a c line like this one.
+  EXPECT_NE(message.find("c=IN IP6 fe80::1234:5678:abcd:ef12"),
+            std::string::npos);
+  // Shouldn't have a IP4 c line.
+  EXPECT_EQ(message.find("c=IN IP4"), std::string::npos);
+}
+
+// This tests serialization of SDP with only TCP candidates and verifies that
+// null IPv4 is used as default address in c line.
+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithTCPOnly) {
+  // Only test 1 m line.
+  desc_.RemoveContentByName("video_content_name");
+  // Stun has a high preference than local host.
+  cricket::Candidate candidate1(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+      rtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
+      cricket::STUN_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  cricket::Candidate candidate2(
+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+      rtc::SocketAddress("::2", 1235), kCandidatePriority, "", "",
+      cricket::LOCAL_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
+  JsepSessionDescription jdesc(kDummyString);
+  ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+  // Only add the candidates to audio m line.
+  JsepIceCandidate jice1("audio_content_name", 0, candidate1);
+  JsepIceCandidate jice2("audio_content_name", 0, candidate2);
+  ASSERT_TRUE(jdesc.AddCandidate(&jice1));
+  ASSERT_TRUE(jdesc.AddCandidate(&jice2));
+  std::string message = webrtc::SdpSerialize(jdesc);
+
+  // Audio line should have a c line like this one when no any default exists.
+  EXPECT_NE(message.find("c=IN IP4 0.0.0.0"), std::string::npos);
+}
+
 // This tests serialization of SDP with a=crypto and a=fingerprint, as would be
 // the case in a DTLS offer.
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithFingerprint) {