diff --git a/webrtc/ringrtc/rffi/src/sdp_observer.cc b/webrtc/ringrtc/rffi/src/sdp_observer.cc
index d60f3d5e7ba..d561dedd1d0 100644
--- a/webrtc/ringrtc/rffi/src/sdp_observer.cc
+++ b/webrtc/ringrtc/rffi/src/sdp_observer.cc
@@ -6,7 +6,7 @@
 #include "rffi/api/sdp_observer_intf.h"
 #include "rffi/src/ptr.h"
 #include "rffi/src/sdp_observer.h"
-#include "third_party/re2/src/re2/re2.h"
+#include <regex>
 
 namespace webrtc {
 namespace rffi {
@@ -29,8 +29,8 @@ void CreateSessionDescriptionObserverRffi::OnSuccess(SessionDescriptionInterface
   // TODO tweak the response a little
   std::string sdp;
   if (session_description->ToString(&sdp)) {
-    static LazyRE2 ssrc_re = {".+urn:ietf:params:rtp-hdrext:ssrc-audio-level.*\r?\n"};
-    RE2::Replace(&sdp, *ssrc_re, "");
+    sdp = std::regex_replace(sdp, std::regex("(a=fmtp:111 ((?!cbr=).)*)\r?\n"), "$1;cbr=1\r\n");
+    sdp = std::regex_replace(sdp, std::regex(".+urn:ietf:params:rtp-hdrext:ssrc-audio-level.*\r?\n"), "");
 
     std::unique_ptr<SessionDescriptionInterface> session_description2 = CreateSessionDescription(session_description->GetType(), sdp);
     delete session_description;
diff --git a/webrtc/ringrtc/rffi/BUILD.gn b/webrtc/ringrtc/rffi/BUILD.gn
index 4564e734e63..341535b0fc7 100644
--- a/webrtc/ringrtc/rffi/BUILD.gn
+++ b/webrtc/ringrtc/rffi/BUILD.gn
@@ -58,7 +58,6 @@ if (is_android) {
       "${android_sdk}:libjingle_peerconnection_jni",
       "${android_sdk}:libjingle_peerconnection_metrics_default_jni",
       "//pc:libjingle_peerconnection",
-      "//third_party/re2",
     ]
     output_extension = "so"
   }
@@ -78,7 +77,6 @@ if (is_ios) {
 
     deps = [
       "//third_party/libyuv",
-      "//third_party/re2",
     ]
   }
 }
@@ -94,7 +92,6 @@ if (is_linux || is_mac || is_win) {
     deps = [
       "//sdk:media_constraints",
       "//media:rtc_simulcast_encoder_adapter",
-      "//third_party/re2",
     ]
   }
 }
diff --git a/webrtc/ringrtc/rffi/api/peer_connection_intf.h b/webrtc/ringrtc/rffi/api/peer_connection_intf.h
index 66958254fed..4cd223beb93 100644
--- a/webrtc/ringrtc/rffi/api/peer_connection_intf.h
+++ b/webrtc/ringrtc/rffi/api/peer_connection_intf.h
@@ -105,6 +105,7 @@ RUSTEXPORT webrtc::SessionDescriptionInterface*
 Rust_sessionDescriptionFromV4(bool offer,
                               const RffiConnectionParametersV4* v4_borrowed,
                               bool enable_tcc_audio,
+                              bool enable_red_audio,
                               bool enable_vp9);
 
 RUSTEXPORT void
diff --git a/webrtc/ringrtc/rffi/src/peer_connection.cc b/webrtc/ringrtc/rffi/src/peer_connection.cc
index 9db5ed8219d..0714b3589e3 100644
--- a/webrtc/ringrtc/rffi/src/peer_connection.cc
+++ b/webrtc/ringrtc/rffi/src/peer_connection.cc
@@ -42,6 +42,7 @@ int VIDEO_LAYERS_ALLOCATION_EXT_ID = 14;
 // 101 used by connection.rs
 int DATA_PT = 101;
 int OPUS_PT = 102;
+int OPUS_RED_PT = 105;
 int VP8_PT = 108;
 int VP8_RTX_PT = 118;
 int VP9_PT = 109;
@@ -317,12 +318,14 @@ RUSTEXPORT webrtc::SessionDescriptionInterface*
 Rust_sessionDescriptionFromV4(bool offer,
                               const RffiConnectionParametersV4* v4_borrowed,
                               bool enable_tcc_audio,
+                              bool enable_red_audio,
                               bool enable_vp9) {
   // Major changes from the default WebRTC behavior:
   // 1. We remove all codecs except Opus, VP8, and VP9
   // 2. We remove all header extensions except for transport-cc, video orientation,
   //    and abs send time.
   // 3. Opus CBR and DTX is enabled.
+  // 4. RED is enabled for audio.
 
   // For some reason, WebRTC insists that the video SSRCs for one side don't 
   // overlap with SSRCs from the other side.  To avoid potential problems, we'll give the
@@ -361,6 +364,15 @@ Rust_sessionDescriptionFromV4(bool offer,
   auto video = std::make_unique<cricket::VideoContentDescription>();
   set_rtp_params(video.get());
 
+  // Turn on the RED "meta codec" for Opus redundancy.
+  auto opus_red = cricket::CreateAudioCodec(OPUS_RED_PT, cricket::kRedCodecName, 48000, 2);
+  opus_red.SetParam("", std::to_string(OPUS_PT) + "/" + std::to_string(OPUS_PT));
+
+  if (enable_red_audio) {
+    // Add RED before Opus to use it by default when sending.
+    audio->AddCodec(opus_red);
+  }
+
   auto opus = cricket::CreateAudioCodec(OPUS_PT, cricket::kOpusCodecName, 48000, 2);
   // These are the current defaults for WebRTC
   // We set them explicitly to avoid having the defaults change on us.
@@ -378,6 +390,11 @@ Rust_sessionDescriptionFromV4(bool offer,
   opus.AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
   audio->AddCodec(opus);
 
+  if (!enable_red_audio) {
+    // Add RED after Opus so that RED packets can at least be decoded properly if received.
+    audio->AddCodec(opus_red);
+  }
+
   auto add_video_feedback_params = [] (cricket::Codec* video_codec) {
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir));
@@ -589,9 +606,16 @@ CreateSessionDescriptionForGroupCall(bool local,
   opus.SetParam("cbr", "1");
   opus.AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
 
+  // Turn on the RED "meta codec" for Opus redundancy.
+  auto opus_red = cricket::CreateAudioCodec(OPUS_RED_PT, cricket::kRedCodecName, 48000, 2);
+  opus_red.SetParam("", std::to_string(OPUS_PT) + "/" + std::to_string(OPUS_PT));
+
+  // Add RED after Opus so that RED packets can at least be decoded properly if received.
   local_audio->AddCodec(opus);
+  local_audio->AddCodec(opus_red);
   for (auto& remote_audio : remote_audios) {
     remote_audio->AddCodec(opus);
+    remote_audio->AddCodec(opus_red);
   }
 
   auto add_video_feedback_params = [] (cricket::Codec* video_codec) {