diff --git a/examples/Cargo.toml b/examples/Cargo.toml
index d95b25cd7..a500f5c3e 100644
--- a/examples/Cargo.toml
+++ b/examples/Cargo.toml
@@ -31,6 +31,11 @@ serde_json = "1.0"
 bytes = "1.1"
 lazy_static = "1.4"
 rand = "0.8"
+# encode-microphone
+cpal = "0.14.0"
+opus = "0.3.0"
+flume = "0.10.14"
+base64 = "0.13.0"
 
 
 [[example]]
@@ -147,3 +152,8 @@ bench = false
 name = "ice-restart"
 path = "examples/ice-restart/ice-restart.rs"
 bench = false
+
+[[example]]
+name = "encode-microphone"
+path = "examples/encode-microphone/encode-microphone.rs"
+bench = false
diff --git a/examples/examples/encode-microphone/README.md b/examples/examples/encode-microphone/README.md
new file mode 100644
index 000000000..a71eda40b
--- /dev/null
+++ b/examples/examples/encode-microphone/README.md
@@ -0,0 +1,29 @@
+# encode-microphone
+encode-microphone demonstrates how to send audio to your browser from your microphone (input device).
+
+## Instructions
+### Build encode-microphone
+```
+cargo build --example encode-microphone
+```
+
+### Open encode-microphone example page
+[jsfiddle.net](https://jsfiddle.net/9s10amwL/) you should see two text-areas and a 'Start Session' button
+
+### Run encode-microphone with your browsers SessionDescription as stdin
+In the jsfiddle the top textarea is your browser, copy that and:
+
+#### Linux/macOS
+Run `echo $BROWSER_SDP | ./target/debug/examples/encode-microphone`
+ 
+#### Windows
+1. Paste the SessionDescription into a file.
+2. Run `./target/debug/examples/encode-microphone < my_file`
+
+### Input encode-microphone's SessionDescription into your browser
+Copy the text that `encode-microphone` just emitted and copy into second text area
+
+### Hit 'Start Session' in jsfiddle, enjoy your video!
+Audio should start playing in your browser below the input boxes.
+
+Congrats, you have used WebRTC.rs!
diff --git a/examples/examples/encode-microphone/encode-microphone.rs b/examples/examples/encode-microphone/encode-microphone.rs
new file mode 100644
index 000000000..0fcd063cb
--- /dev/null
+++ b/examples/examples/encode-microphone/encode-microphone.rs
@@ -0,0 +1,338 @@
+use anyhow::Result;
+use bytes::Bytes;
+use clap::{AppSettings, Arg, Command};
+use cpal::traits::DeviceTrait;
+use cpal::traits::HostTrait;
+use cpal::traits::StreamTrait;
+use cpal::Device;
+use cpal::DevicesError;
+use cpal::SampleRate;
+use std::io::Write;
+use std::sync::Arc;
+use std::thread;
+use tokio::sync::Notify;
+use tokio::time::Duration;
+use webrtc::api::interceptor_registry::register_default_interceptors;
+use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS};
+use webrtc::api::APIBuilder;
+use webrtc::ice_transport::ice_connection_state::RTCIceConnectionState;
+use webrtc::ice_transport::ice_server::RTCIceServer;
+use webrtc::interceptor::registry::Registry;
+use webrtc::media::Sample;
+use webrtc::peer_connection::configuration::RTCConfiguration;
+use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
+use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
+use webrtc::rtp_transceiver::rtp_codec::RTCRtpCodecCapability;
+use webrtc::track::track_local::track_local_static_sample::TrackLocalStaticSample;
+use webrtc::track::track_local::TrackLocal;
+
+#[tokio::main]
+async fn main() -> Result<()> {
+    let mut app = Command::new("encode-microphone")
+        .version("0.1.0")
+        .author("Mohamad Rajabi <mo.rajbi@gmail.com>")
+        .about("An example of getting mic stream and encoding audio using opus.")
+        .setting(AppSettings::DeriveDisplayOrder)
+        .subcommand_negates_reqs(true)
+        .arg(
+            Arg::new("FULLHELP")
+                .help("Prints more detailed help information")
+                .long("fullhelp"),
+        )
+        .arg(
+            Arg::new("debug")
+                .long("debug")
+                .short('d')
+                .help("Prints debug log information"),
+        );
+
+    let matches = app.clone().get_matches();
+
+    if matches.is_present("FULLHELP") {
+        app.print_long_help().unwrap();
+        std::process::exit(0);
+    }
+
+    let debug = matches.is_present("debug");
+    if debug {
+        env_logger::Builder::new()
+            .format(|buf, record| {
+                writeln!(
+                    buf,
+                    "{}:{} [{}] {} - {}",
+                    record.file().unwrap_or("unknown"),
+                    record.line().unwrap_or(0),
+                    record.level(),
+                    chrono::Local::now().format("%H:%M:%S.%6f"),
+                    record.args()
+                )
+            })
+            .filter(None, log::LevelFilter::Trace)
+            .init();
+    }
+
+    // Everything below is the WebRTC-rs API! Thanks for using it ❤️.
+
+    // Create a MediaEngine object to configure the supported codec
+    let mut m = MediaEngine::default();
+
+    m.register_default_codecs()?;
+
+    // Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
+    // This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
+    // this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
+    // for each PeerConnection.
+    let mut registry = Registry::new();
+
+    // Use the default set of Interceptors
+    registry = register_default_interceptors(registry, &mut m)?;
+
+    // Create the API object with the MediaEngine
+    let api = APIBuilder::new()
+        .with_media_engine(m)
+        .with_interceptor_registry(registry)
+        .build();
+
+    // Prepare the configuration
+    let config = RTCConfiguration {
+        ice_servers: vec![RTCIceServer {
+            urls: vec!["stun:stun.l.google.com:19302".to_owned()],
+            ..Default::default()
+        }],
+        ..Default::default()
+    };
+
+    // Create a new RTCPeerConnection
+    let peer_connection = Arc::new(api.new_peer_connection(config).await?);
+
+    let notify_tx = Arc::new(Notify::new());
+    let notify_audio = notify_tx.clone();
+
+    let (done_tx, mut done_rx) = tokio::sync::mpsc::channel::<()>(1);
+
+    // Create a audio track
+    let audio_track = Arc::new(TrackLocalStaticSample::new(
+        RTCRtpCodecCapability {
+            mime_type: MIME_TYPE_OPUS.to_owned(),
+            ..Default::default()
+        },
+        "audio".to_owned(),
+        "webrtc-rs".to_owned(),
+    ));
+
+    // Add this newly created track to the PeerConnection
+    let rtp_sender = peer_connection
+        .add_track(Arc::clone(&audio_track) as Arc<dyn TrackLocal + Send + Sync>)
+        .await?;
+
+    // Read incoming RTCP packets
+    // Before these packets are returned they are processed by interceptors. For things
+    // like NACK this needs to be called.
+    tokio::spawn(async move {
+        let mut rtcp_buf = vec![0u8; 1500];
+        while let Ok((_, _)) = rtp_sender.read(&mut rtcp_buf).await {}
+        Result::<()>::Ok(())
+    });
+
+    let (sender, frame_receiver) = flume::bounded::<AudioFrame>(3);
+    let (encoded_sender, encoded_receiver) = flume::bounded::<AudioEncodedFrame>(3);
+
+    // Encoder thread
+    thread::spawn(move || {
+        // We just handle 48khz, to handle other sample rates like 44.1khz you need to use a resampler.
+        let mut encoder =
+            opus::Encoder::new(48000, opus::Channels::Mono, opus::Application::Voip).unwrap();
+
+        loop {
+            let AudioFrame { data } = frame_receiver.recv().unwrap();
+
+            let sample_count = data.len() as u64;
+            // sample duration
+            let duration = Duration::from_millis(sample_count * 1000 / 48000);
+            let encoded = encoder
+                .encode_vec_float(&data, 1024)
+                .expect("Failed to encode");
+            let bytes = Bytes::from(encoded);
+
+            encoded_sender
+                .send(AudioEncodedFrame { bytes, duration })
+                .unwrap();
+        }
+    });
+
+    // STREAM
+    let device = get_default_input_device().expect("Failed to get default device.");
+
+    // ---
+    let input_configs = match device.supported_input_configs() {
+        Ok(f) => f,
+        Err(e) => {
+            panic!("Error getting supported input configs: {:?}", e);
+        }
+    };
+    let input_configs2 = input_configs
+        .into_iter()
+        .find(|c| c.max_sample_rate() == SampleRate(48000))
+        .expect("did not find a sample rate of 48khz");
+
+    let config = input_configs2.with_sample_rate(SampleRate(48000));
+
+    let err_fn = move |err| {
+        eprintln!("an error occurred on stream: {}", err);
+    };
+
+    // until it is 960
+    let mut buffer: Vec<f32> = Vec::new();
+
+    // assume cpal::SampleFormat::F32
+    let stream = device
+        .build_input_stream(
+            &config.into(),
+            move |data: &[f32], _| {
+                for &sample in data {
+                    buffer.push(sample.clone());
+                    if buffer.len() == 960 {
+                        sender
+                            .send(AudioFrame {
+                                data: Arc::new(buffer.to_owned()),
+                            })
+                            .expect("Failed to send raw frame to the encoder");
+                        // Create a new vec
+                        buffer.clear();
+                    }
+                }
+            },
+            err_fn,
+        )
+        .unwrap();
+
+    stream.play().unwrap();
+
+    // SENDER
+    tokio::spawn(async move {
+        // Wait for connection established
+        let _ = notify_audio.notified().await;
+
+        println!("send the audio from the encoder");
+
+        while let Ok(frame) = encoded_receiver.recv_async().await {
+            // frame
+            audio_track
+                .write_sample(&Sample {
+                    data: frame.bytes,
+                    duration: frame.duration,
+                    ..Default::default()
+                })
+                .await?;
+        }
+
+        Result::<()>::Ok(())
+    });
+
+    // Set the handler for ICE connection state
+    // This will notify you when the peer has connected/disconnected
+    peer_connection
+        .on_ice_connection_state_change(Box::new(move |connection_state: RTCIceConnectionState| {
+            println!("Connection State has changed {}", connection_state);
+            if connection_state == RTCIceConnectionState::Connected {
+                notify_tx.notify_waiters();
+            }
+            Box::pin(async {})
+        }))
+        .await;
+
+    // Set the handler for Peer connection state
+    // This will notify you when the peer has connected/disconnected
+    peer_connection
+        .on_peer_connection_state_change(Box::new(move |s: RTCPeerConnectionState| {
+            println!("Peer Connection State has changed: {}", s);
+
+            if s == RTCPeerConnectionState::Failed {
+                // Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
+                // Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
+                // Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
+                println!("Peer Connection has gone to failed exiting");
+                let _ = done_tx.try_send(());
+            }
+
+            Box::pin(async {})
+        }))
+        .await;
+
+    // Wait for the offer to be pasted
+    let line = signal::must_read_stdin()?;
+    let desc_data = signal::decode(line.as_str())?;
+    let offer = serde_json::from_str::<RTCSessionDescription>(&desc_data)?;
+
+    // Set the remote SessionDescription
+    peer_connection.set_remote_description(offer).await?;
+
+    // Create an answer
+    let answer = peer_connection.create_answer(None).await?;
+
+    // Create channel that is blocked until ICE Gathering is complete
+    let mut gather_complete = peer_connection.gathering_complete_promise().await;
+
+    // Sets the LocalDescription, and starts our UDP listeners
+    peer_connection.set_local_description(answer).await?;
+
+    // Block until ICE Gathering is complete, disabling trickle ICE
+    // we do this because we only can exchange one signaling message
+    // in a production application you should exchange ICE Candidates via OnICECandidate
+    let _ = gather_complete.recv().await;
+
+    // Output the answer in base64 so we can paste it in browser
+    if let Some(local_desc) = peer_connection.local_description().await {
+        let json_str = serde_json::to_string(&local_desc)?;
+        let b64 = signal::encode(&json_str);
+        println!("{}", b64);
+    } else {
+        println!("generate local_description failed!");
+    }
+
+    println!("Press ctrl-c to stop");
+    tokio::select! {
+        _ = done_rx.recv() => {
+            println!("received done signal!");
+        }
+        _ = tokio::signal::ctrl_c() => {
+            println!("");
+        }
+    };
+
+    peer_connection.close().await?;
+
+    drop(stream);
+
+    Ok(())
+}
+
+fn get_default_input_device() -> Result<Device, DevicesError> {
+    let device = "default";
+
+    #[cfg(any(
+        not(any(target_os = "linux", target_os = "dragonfly", target_os = "freebsd")),
+        not(feature = "jack")
+    ))]
+    let host = cpal::default_host();
+
+    // Set up the input device and stream with the default input config.
+    let device = if device == "default" {
+        host.default_input_device()
+    } else {
+        host.input_devices()?
+            .find(|x| x.name().map(|y| y == device).unwrap_or(false))
+    }
+    .expect("failed to find input device");
+
+    Ok(device)
+}
+
+struct AudioFrame {
+    data: Arc<Vec<f32>>,
+}
+
+struct AudioEncodedFrame {
+    bytes: Bytes,
+    duration: Duration,
+}