diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index 129eb3288a..335ee3a691 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -1560,6 +1560,11 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
 
+  // ValidateStreamParams should prevent this from happening.
+  RTC_CHECK(!parameters_.config.rtp.ssrcs.empty());
+  rtp_parameters_.encodings[0].ssrc =
+      rtc::Optional<uint32_t>(parameters_.config.rtp.ssrcs[0]);
+
   // RTX.
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
@@ -1862,11 +1867,16 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::GetRtpParameters() const {
 
 bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters(
     const webrtc::RtpParameters& rtp_parameters) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   if (rtp_parameters.encodings.size() != 1) {
     LOG(LS_ERROR)
         << "Attempted to set RtpParameters without exactly one encoding";
     return false;
   }
+  if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
+    LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
+    return false;
+  }
   return true;
 }
 
diff --git a/webrtc/media/engine/webrtcvideoengine2_unittest.cc b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
index b8495b0546..5944883e1c 100644
--- a/webrtc/media/engine/webrtcvideoengine2_unittest.cc
+++ b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
@@ -3661,16 +3661,21 @@ TEST_F(WebRtcVideoChannel2Test,
   // for each encoding individually.
 
   AddSendStream();
-  // Setting RtpParameters with no encoding is expected to fail.
   webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
-  parameters.encodings.clear();
-  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
-  // Setting RtpParameters with exactly one encoding should succeed.
-  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
   // Two or more encodings should result in failure.
   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
   EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
+  // Zero encodings should also fail.
+  parameters.encodings.clear();
+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
+}
+
+// Changing the SSRC through RtpParameters is not allowed.
+TEST_F(WebRtcVideoChannel2Test, CannotSetSsrcInRtpSendParameters) {
+  AddSendStream();
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
+  parameters.encodings[0].ssrc = rtc::Optional<uint32_t>(0xdeadbeef);
+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
 }
 
 // Test that a stream will not be sending if its encoding is made inactive
@@ -3753,6 +3758,18 @@ TEST_F(WebRtcVideoChannel2Test, GetRtpSendParametersCodecs) {
             rtp_parameters.codecs[1]);
 }
 
+// Test that RtpParameters for send stream has one encoding and it has
+// the correct SSRC.
+TEST_F(WebRtcVideoChannel2Test, GetRtpSendParametersSsrc) {
+  AddSendStream();
+
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpSendParameters(last_ssrc_);
+  ASSERT_EQ(1u, rtp_parameters.encodings.size());
+  EXPECT_EQ(rtc::Optional<uint32_t>(last_ssrc_),
+            rtp_parameters.encodings[0].ssrc);
+}
+
 // Test that if we set/get parameters multiple times, we get the same results.
 TEST_F(WebRtcVideoChannel2Test, SetAndGetRtpSendParameters) {
   AddSendStream();
@@ -3826,7 +3843,7 @@ TEST_F(WebRtcVideoChannel2Test, DISABLED_GetRtpReceiveFmtpSprop) {
 
 // Test that RtpParameters for receive stream has one encoding and it has
 // the correct SSRC.
-TEST_F(WebRtcVideoChannel2Test, RtpEncodingParametersSsrcIsSet) {
+TEST_F(WebRtcVideoChannel2Test, GetRtpReceiveParametersSsrc) {
   AddRecvStream();
 
   webrtc::RtpParameters rtp_parameters =
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index a36f328705..9871d3a21f 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1384,8 +1384,23 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     return rtp_parameters_;
   }
 
+  bool ValidateRtpParameters(const webrtc::RtpParameters& rtp_parameters) {
+    if (rtp_parameters.encodings.size() != 1) {
+      LOG(LS_ERROR)
+          << "Attempted to set RtpParameters without exactly one encoding";
+      return false;
+    }
+    if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
+      LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
+      return false;
+    }
+    return true;
+  }
+
   bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
-    RTC_CHECK_EQ(1UL, parameters.encodings.size());
+    if (!ValidateRtpParameters(parameters)) {
+      return false;
+    }
     auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
                                         parameters.encodings[0].max_bitrate_bps,
                                         send_codec_spec_.codec_inst);
@@ -1704,9 +1719,6 @@ bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  if (!ValidateRtpParameters(parameters)) {
-    return false;
-  }
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
@@ -1760,9 +1772,6 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  if (!ValidateRtpParameters(parameters)) {
-    return false;
-  }
   auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
@@ -1779,16 +1788,6 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
   return true;
 }
 
-bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
-    const webrtc::RtpParameters& rtp_parameters) {
-  if (rtp_parameters.encodings.size() != 1) {
-    LOG(LS_ERROR)
-        << "Attempted to set RtpParameters without exactly one encoding";
-    return false;
-  }
-  return true;
-}
-
 bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "Setting voice channel options: "
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 3ca7ec8b05..0cfbeab42b 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -1039,15 +1039,21 @@ TEST_F(WebRtcVoiceEngineTestFake,
   // for each encoding individually.
 
   EXPECT_TRUE(SetupSendStream());
-  // Setting RtpParameters with no encoding is expected to fail.
-  webrtc::RtpParameters parameters;
-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
-  // Setting RtpParameters with exactly one encoding should succeed.
-  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrc1);
   // Two or more encodings should result in failure.
   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
   EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+  // Zero encodings should also fail.
+  parameters.encodings.clear();
+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+}
+
+// Changing the SSRC through RtpParameters is not allowed.
+TEST_F(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) {
+  EXPECT_TRUE(SetupSendStream());
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrc1);
+  parameters.encodings[0].ssrc = rtc::Optional<uint32_t>(0xdeadbeef);
+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
 }
 
 // Test that a stream will not be sending if its encoding is made