diff --git a/src/audio_utils.hpp b/src/audio_utils.hpp
index cbe9dae..07e66f7 100644
--- a/src/audio_utils.hpp
+++ b/src/audio_utils.hpp
@@ -43,7 +43,10 @@ struct StreamParams {
     std::string device_name;
     int sample_rate;
     int num_channels;
-    double latency_seconds;
+    // Hint passed to PortAudio when opening the stream. PortAudio may choose the closest
+    // viable latency instead, rounding up to the next practical value.
+    // Use get_stream_latency() after opening the stream to retrieve the actual latency PortAudio selected.
+    double suggested_latency_seconds;
     bool is_input;
     PaStreamCallback* callback;
     void* user_data;  // Points to AudioStreamContext* or PlaybackBuffer*
@@ -171,9 +174,9 @@ inline StreamParams setupStreamFromConfig(const ConfigParams& params,
     const double default_latency =
         (direction == StreamDirection::Input) ? deviceInfo->defaultLowInputLatency : deviceInfo->defaultLowOutputLatency;
 
-    stream_params.latency_seconds = params.latency_ms.has_value() ? params.latency_ms.value() / 1000.0 : default_latency;
+    stream_params.suggested_latency_seconds = params.latency_ms.has_value() ? params.latency_ms.value() / 1000.0 : default_latency;
 
-    VIAM_SDK_LOG(debug) << "[setupStreamFromConfig] Using latency " << stream_params.latency_seconds;
+    VIAM_SDK_LOG(debug) << "[setupStreamFromConfig] Using latency " << stream_params.suggested_latency_seconds;
 
     // Validate num_channels against device's max channels
     const int max_channels = (direction == StreamDirection::Input) ? deviceInfo->maxInputChannels : deviceInfo->maxOutputChannels;
@@ -202,7 +205,7 @@ inline void openStream(PaStream*& stream, const StreamParams& params, const audi
     stream_params.device = params.device_index;
     stream_params.channelCount = params.num_channels;
     stream_params.sampleFormat = paInt16;
-    stream_params.suggestedLatency = params.latency_seconds;
+    stream_params.suggestedLatency = params.suggested_latency_seconds;
     stream_params.hostApiSpecificStreamInfo = nullptr;
 
     // Determine which parameter is input and which is output
@@ -218,7 +221,7 @@ inline void openStream(PaStream*& stream, const StreamParams& params, const audi
                << "  - Sample rate: " << params.sample_rate << " Hz\n"
                << "  - Channels: " << params.num_channels << "\n"
                << "  - Format: 16-bit PCM\n"
-               << "  - Latency: " << params.latency_seconds << " seconds";
+               << "  - Latency: " << params.suggested_latency_seconds << " seconds";
         VIAM_SDK_LOG(error) << buffer.str();
         throw std::runtime_error(buffer.str());
     }
@@ -280,6 +283,18 @@ inline void shutdown_stream(PaStream* stream, const audio::portaudio::PortAudioI
     }
 }
 
+// Returns the actual latency reported by PortAudio after stream open (inputLatency for
+// input streams, outputLatency for output streams). Falls back to suggested_latency_seconds
+// if stream info is unavailable.
+inline double get_stream_latency(PaStream* stream, const StreamParams& params, const audio::portaudio::PortAudioInterface* pa = nullptr) {
+    audio::portaudio::RealPortAudio real_pa;
+    const audio::portaudio::PortAudioInterface& audio_interface = pa ? *pa : real_pa;
+    const PaStreamInfo* stream_info = audio_interface.getStreamInfo(stream);
+    if (!stream_info)
+        return params.suggested_latency_seconds;
+    return params.is_input ? stream_info->inputLatency : stream_info->outputLatency;
+}
+
 inline void restart_stream(PaStream*& stream, const StreamParams& params, const audio::portaudio::PortAudioInterface* pa = nullptr) {
     // In production pa is nullptr and real_pa is used. For testing, pa is the mock pa
     audio::portaudio::RealPortAudio real_pa;
diff --git a/src/microphone.cpp b/src/microphone.cpp
index 556b1ae..923b839 100644
--- a/src/microphone.cpp
+++ b/src/microphone.cpp
@@ -157,11 +157,11 @@ Microphone::Microphone(viam::sdk::Dependencies deps, viam::sdk::ResourceConfig c
         requested_sample_rate_ =
             setup.config_params.sample_rate.value_or(setup.stream_params.sample_rate);  // User's requested rate, defaults to device rate
         num_channels_ = setup.stream_params.num_channels;
-        latency_ = setup.stream_params.latency_seconds;
         audio_context_ = setup.audio_context;
         historical_throttle_ms_ = setup.config_params.historical_throttle_ms.value_or(DEFAULT_HISTORICAL_THROTTLE_MS);
 
         audio::utils::restart_stream(stream_, setup.stream_params, pa_);
+        latency_ = audio::utils::get_stream_latency(stream_, setup.stream_params, pa_);
     }
 }
 
@@ -275,7 +275,7 @@ void Microphone::reconfigure(const viam::sdk::Dependencies& deps, const viam::sd
             requested_sample_rate_ = setup.config_params.sample_rate.value_or(
                 setup.stream_params.sample_rate);  // User's requested rate, defaults to device rate
             num_channels_ = setup.stream_params.num_channels;
-            latency_ = setup.stream_params.latency_seconds;
+            latency_ = audio::utils::get_stream_latency(stream_, setup.stream_params, pa_);
             audio_context_ = setup.audio_context;
             historical_throttle_ms_ = setup.config_params.historical_throttle_ms.value_or(DEFAULT_HISTORICAL_THROTTLE_MS);
         }
diff --git a/src/speaker.cpp b/src/speaker.cpp
index c408ea5..8d8e54e 100644
--- a/src/speaker.cpp
+++ b/src/speaker.cpp
@@ -30,9 +30,9 @@ Speaker::Speaker(viam::sdk::Dependencies deps, viam::sdk::ResourceConfig cfg, au
         device_name_ = setup.stream_params.device_name;
         sample_rate_ = setup.stream_params.sample_rate;
         num_channels_ = setup.stream_params.num_channels;
-        latency_ = setup.stream_params.latency_seconds;
         audio_context_ = setup.audio_context;
         audio::utils::restart_stream(stream_, setup.stream_params, pa_);
+        latency_ = audio::utils::get_stream_latency(stream_, setup.stream_params, pa_);
         volume_ = setup.config_params.volume;
         if (volume_) {
             audio::volume::set_volume(device_name_, *volume_);
@@ -390,7 +390,7 @@ void Speaker::reconfigure(const vsdk::Dependencies& deps, const vsdk::ResourceCo
             device_name_ = setup.stream_params.device_name;
             sample_rate_ = setup.stream_params.sample_rate;
             num_channels_ = setup.stream_params.num_channels;
-            latency_ = setup.stream_params.latency_seconds;
+            latency_ = audio::utils::get_stream_latency(stream_, setup.stream_params, pa_);
             audio_context_ = setup.audio_context;
             volume_ = setup.config_params.volume;
             if (volume_) {
diff --git a/test/audio_utils_test.cpp b/test/audio_utils_test.cpp
index 2fc3022..9cd006d 100644
--- a/test/audio_utils_test.cpp
+++ b/test/audio_utils_test.cpp
@@ -162,7 +162,7 @@ TEST_F(AudioUtilsTest, SetupStreamFromConfigUsesProvidedValues) {
 
     EXPECT_EQ(stream_params.sample_rate, 44100);
     EXPECT_EQ(stream_params.num_channels, 2);
-    EXPECT_DOUBLE_EQ(stream_params.latency_seconds, 0.1);
+    EXPECT_DOUBLE_EQ(stream_params.suggested_latency_seconds, 0.1);
 }
 
 TEST_F(AudioUtilsTest, SetupStreamFromConfigOutputDirection) {
@@ -364,6 +364,48 @@ TEST_F(AudioUtilsTest, SetupAudioDeviceUsesConfigParams) {
     EXPECT_EQ(setup.config_params.device_name, "My Device");
 }
 
+TEST_F(AudioUtilsTest, GetStreamLatencyFallsBackWhenStreamInfoNull) {
+    using ::testing::Return;
+    ON_CALL(*mock_pa_, getStreamInfo(::testing::_)).WillByDefault(Return(nullptr));
+
+    audio::utils::StreamParams params;
+    params.suggested_latency_seconds = 0.05;
+    params.is_input = false;
+
+    double latency = audio::utils::get_stream_latency(nullptr, params, mock_pa_.get());
+    EXPECT_DOUBLE_EQ(latency, 0.05);
+}
+
+TEST_F(AudioUtilsTest, GetStreamLatencyReturnsOutputLatency) {
+    using ::testing::Return;
+    PaStreamInfo stream_info;
+    stream_info.inputLatency  = 0.02;
+    stream_info.outputLatency = 0.04;
+    ON_CALL(*mock_pa_, getStreamInfo(::testing::_)).WillByDefault(Return(&stream_info));
+
+    audio::utils::StreamParams params;
+    params.suggested_latency_seconds = 0.01;
+    params.is_input = false;
+
+    double latency = audio::utils::get_stream_latency(nullptr, params, mock_pa_.get());
+    EXPECT_DOUBLE_EQ(latency, 0.04);
+}
+
+TEST_F(AudioUtilsTest, GetStreamLatencyReturnsInputLatency) {
+    using ::testing::Return;
+    PaStreamInfo stream_info;
+    stream_info.inputLatency  = 0.02;
+    stream_info.outputLatency = 0.04;
+    ON_CALL(*mock_pa_, getStreamInfo(::testing::_)).WillByDefault(Return(&stream_info));
+
+    audio::utils::StreamParams params;
+    params.suggested_latency_seconds = 0.01;
+    params.is_input = true;
+
+    double latency = audio::utils::get_stream_latency(nullptr, params, mock_pa_.get());
+    EXPECT_DOUBLE_EQ(latency, 0.02);
+}
+
 int main(int argc, char **argv) {
   ::testing::InitGoogleTest(&argc, argv);
   ::testing::AddGlobalTestEnvironment(new test_utils::AudioTestEnvironment);