diff --git a/media/base/mediaengine.cc b/media/base/mediaengine.cc
index 93d72c2e34..d85e12773f 100644
--- a/media/base/mediaengine.cc
+++ b/media/base/mediaengine.cc
@@ -19,4 +19,18 @@ webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
   return parameters;
 }
 
+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
+  std::vector<uint32_t> primary_ssrcs;
+  sp.GetPrimarySsrcs(&primary_ssrcs);
+  size_t encoding_count = primary_ssrcs.size();
+
+  std::vector<webrtc::RtpEncodingParameters> encodings(encoding_count);
+  for (size_t i = 0; i < encodings.size(); ++i) {
+    encodings[i].ssrc = primary_ssrcs[i];
+  }
+  webrtc::RtpParameters parameters;
+  parameters.encodings = encodings;
+  return parameters;
+}
+
 };  // namespace cricket
diff --git a/media/base/mediaengine.h b/media/base/mediaengine.h
index fcb4ad97b7..cb0d37ddfd 100644
--- a/media/base/mediaengine.h
+++ b/media/base/mediaengine.h
@@ -160,6 +160,7 @@ class DataEngineInterface {
 };
 
 webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
 
 }  // namespace cricket
 
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 411904f8e8..8be1f7b956 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1557,7 +1557,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
       stream_(nullptr),
       encoder_sink_(nullptr),
       parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
-      rtp_parameters_(CreateRtpParametersWithOneEncoding()),
+      rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
       sending_(false) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
   parameters_.conference_mode = send_params.conference_mode;
@@ -1814,9 +1814,9 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetRtpParameters() const {
 bool WebRtcVideoChannel::WebRtcVideoSendStream::ValidateRtpParameters(
     const webrtc::RtpParameters& rtp_parameters) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (rtp_parameters.encodings.size() != 1) {
+  if (rtp_parameters.encodings.size() != rtp_parameters_.encodings.size()) {
     RTC_LOG(LS_ERROR)
-        << "Attempted to set RtpParameters without exactly one encoding";
+        << "Attempted to set RtpParameters with different encoding count";
     return false;
   }
   if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
@@ -1833,8 +1833,7 @@ bool WebRtcVideoChannel::WebRtcVideoSendStream::ValidateRtpParameters(
 
 void WebRtcVideoChannel::WebRtcVideoSendStream::UpdateSendState() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
-  // TODO(deadbeef): Need to handle more than one encoding in the future.
-  RTC_DCHECK(rtp_parameters_.encodings.size() == 1u);
+  // TODO(zstein): Handle multiple encodings.
   if (sending_ && rtp_parameters_.encodings[0].active) {
     RTC_DCHECK(stream_ != nullptr);
     stream_->Start();
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index 82154a8c0d..31f6d0eaf0 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -4364,11 +4364,6 @@ TEST_F(WebRtcVideoChannelTest, CannotSetMaxBitrateForNonexistentStream) {
 
 TEST_F(WebRtcVideoChannelTest,
        CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
-  // This test verifies that setting RtpParameters succeeds only if
-  // the structure contains exactly one encoding.
-  // TODO(skvlad): Update this test when we start supporting setting parameters
-  // for each encoding individually.
-
   AddSendStream();
   webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
   // Two or more encodings should result in failure.
@@ -4379,6 +4374,22 @@ TEST_F(WebRtcVideoChannelTest,
   EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
 }
 
+TEST_F(WebRtcVideoChannelTest,
+       CannotSetSimulcastRtpSendParametersWithIncorrectNumberOfEncodings) {
+  std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
+  StreamParams sp = CreateSimStreamParams("cname", ssrcs);
+  AddSendStream(sp);
+
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
+
+  // Additional 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(WebRtcVideoChannelTest, CannotSetSsrcInRtpSendParameters) {
   AddSendStream();
@@ -4464,7 +4475,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
   // Get and set the rtp encoding parameters.
   webrtc::RtpParameters parameters =
       channel_->GetRtpSendParameters(primary_ssrc);
-  EXPECT_EQ(1UL, parameters.encodings.size());
+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
   EXPECT_EQ(webrtc::kDefaultBitratePriority,
             parameters.encodings[0].bitrate_priority);
   // Change the value and set it on the VideoChannel.
@@ -4474,7 +4485,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
 
   // Verify that the encoding parameters priority is set on the VideoChannel.
   parameters = channel_->GetRtpSendParameters(primary_ssrc);
-  EXPECT_EQ(1UL, parameters.encodings.size());
+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
   EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority);
 
   // Verify that the new value propagated down to the encoder.
@@ -4505,8 +4516,8 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
 
 // Test that a stream will not be sending if its encoding is made inactive
 // through SetRtpSendParameters.
-// TODO(deadbeef): Update this test when we start supporting setting parameters
-// for each encoding individually.
+// TODO(bugs.webrtc.org/8653): Update this test when we support active/inactive
+// for individual encodings.
 TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersEncodingsActive) {
   FakeVideoSendStream* stream = AddSendStream();
   EXPECT_TRUE(channel_->SetSend(true));
@@ -4845,6 +4856,9 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
     channel_->SetSend(true);
     EXPECT_TRUE(capturer.CaptureFrame());
 
+    auto rtp_parameters = channel_->GetRtpSendParameters(kSsrcs3[0]);
+    EXPECT_EQ(num_configured_streams, rtp_parameters.encodings.size());
+
     std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
     ASSERT_EQ(expected_num_streams, video_streams.size());