diff --git a/python-webrtc/cpp/src/interfaces/rtc_peer_connection.cpp b/python-webrtc/cpp/src/interfaces/rtc_peer_connection.cpp
index 1ea15b8..07562cc 100644
--- a/python-webrtc/cpp/src/interfaces/rtc_peer_connection.cpp
+++ b/python-webrtc/cpp/src/interfaces/rtc_peer_connection.cpp
@@ -81,6 +81,9 @@ namespace python_webrtc {
         .def("getTransceivers", &RTCPeerConnection::GetTransceivers)
         .def("getSenders", &RTCPeerConnection::GetSenders)
         .def("getReceivers", &RTCPeerConnection::GetReceivers)
+        .def_property_readonly("sctp", &RTCPeerConnection::GetSctp)
+        .def("restartIce", &RTCPeerConnection::RestartIce)
+        .def("removeTrack", &RTCPeerConnection::RemoveTrack)
         .def("close", &RTCPeerConnection::Close);
   }
 
@@ -285,6 +288,35 @@ namespace python_webrtc {
     return receivers;
   }
 
+  std::optional<RTCSctpTransport *> RTCPeerConnection::GetSctp() {
+    if (_jinglePeerConnection && _jinglePeerConnection->GetSctpTransport()) {
+      return RTCSctpTransport::holder()->GetOrCreate(_factory, _jinglePeerConnection->GetSctpTransport());
+    }
+
+    return {};
+  }
+
+  void RTCPeerConnection::RestartIce() {
+    if (_jinglePeerConnection) {
+      _jinglePeerConnection->RestartIce();
+    }
+  }
+
+  void RTCPeerConnection::RemoveTrack(RTCRtpSender &sender) {
+    if (!_jinglePeerConnection) {
+      throw PythonWebRTCException("Cannot remove track; RTCPeerConnection is closed");
+    }
+
+    auto senders = _jinglePeerConnection->GetSenders();
+    if (std::find(senders.begin(), senders.end(), sender.sender()) == senders.end()) {
+      throw PythonWebRTCException("Cannot remove track because sender not found in senders of PeerConnection");
+    }
+
+    if (!_jinglePeerConnection->RemoveTrack(sender.sender())) {
+      throw PythonWebRTCException("Cannot remove track");
+    }
+  }
+
   void RTCPeerConnection::Close() {
     if (_jinglePeerConnection) {
       _jinglePeerConnection->Close();
diff --git a/python-webrtc/cpp/src/interfaces/rtc_peer_connection.h b/python-webrtc/cpp/src/interfaces/rtc_peer_connection.h
index b4e31c5..1161f5a 100644
--- a/python-webrtc/cpp/src/interfaces/rtc_peer_connection.h
+++ b/python-webrtc/cpp/src/interfaces/rtc_peer_connection.h
@@ -20,6 +20,7 @@
 #include "media_stream.h"
 #include "rtc_rtp_sender.h"
 #include "rtc_rtp_transceiver.h"
+#include "rtc_sctp_transport.h"
 
 namespace webrtc {
   struct PeerConnectionDependencies;
@@ -65,6 +66,12 @@ namespace python_webrtc {
 
     std::vector<RTCRtpReceiver *> GetReceivers();
 
+    std::optional<RTCSctpTransport *> GetSctp();
+
+    void RestartIce();
+
+    void RemoveTrack(RTCRtpSender &);
+
     void SaveLastSdp(const RTCSessionDescriptionInit &lastSdp);
 
     void Close();
diff --git a/python-webrtc/python/test.py b/python-webrtc/python/test.py
index ebb0029..41d7f0e 100644
--- a/python-webrtc/python/test.py
+++ b/python-webrtc/python/test.py
@@ -59,23 +59,30 @@ async def main():
     # transport = sender.transport
     # transceivers = pc.get_transceivers()
 
-    params = webrtc.RtpEncodingParameters(
-        max_bitrate=1234, max_framerate=20, rid="lolkek", scale_resolution_down_by=2.0
-    )
-    init = webrtc.RtpTransceiverInit(
-        direction=webrtc.TransceiverDirection.recvonly, send_encodings=[params], streams=[stream]
-    )
+    # params = webrtc.RtpEncodingParameters(
+    #     max_bitrate=1234, max_framerate=20, rid="lolkek", scale_resolution_down_by=2.0
+    # )
+    # init = webrtc.RtpTransceiverInit(
+    #     direction=webrtc.TransceiverDirection.recvonly, send_encodings=[params], streams=[stream]
+    # )
+    #
+    # transceiver = pc.add_transceiver(webrtc.MediaType.audio)
+    # transceiver_with_params = pc.add_transceiver(webrtc.MediaType.audio, init)
+    #
+    # transceiver_by_track = pc.add_transceiver(stream.get_tracks()[0])
+    # transceiver_by_track_with_params = pc.add_transceiver(stream.get_tracks()[0], init)
+    #
+    # local_sdp = await pc.create_offer()
+    # await pc.set_local_description(local_sdp)
 
-    transceiver = pc.add_transceiver(webrtc.MediaType.audio)
-    transceiver_with_params = pc.add_transceiver(webrtc.MediaType.audio, init)
+    transceivers = pc.get_transceivers()
 
-    transceiver_by_track = pc.add_transceiver(stream.get_tracks()[0])
-    transceiver_by_track_with_params = pc.add_transceiver(stream.get_tracks()[0], init)
+    pc.restart_ice()
 
-    local_sdp = await pc.create_offer()
-    await pc.set_local_description(local_sdp)
+    sender = pc.add_track(stream.get_tracks()[0], stream)
+    pc.remove_track(sender)
 
-    transceivers = pc.get_transceivers()
+    print(pc.sctp)
 
     idle()
 
diff --git a/python-webrtc/python/webrtc/interfaces/rtc_peer_connection.py b/python-webrtc/python/webrtc/interfaces/rtc_peer_connection.py
index 27ca490..0cf964f 100644
--- a/python-webrtc/python/webrtc/interfaces/rtc_peer_connection.py
+++ b/python-webrtc/python/webrtc/interfaces/rtc_peer_connection.py
@@ -185,10 +185,35 @@ def get_receivers(self) -> List['webrtc.RTCRtpReceiver']:
 
         return RTCRtpReceiver._wrap_many(self._native_obj.getReceivers())
 
+    def remove_track(self, sender: 'webrtc.RTCRtpSender') -> None:
+        """Tells the local end of the connection to stop sending media from the specified track, without
+        actually removing the corresponding :obj:`webrtc.RTCRtpSender` from the list of senders as reported
+        by :attr:`get_senders``. If the track is already stopped, or is not in the connection's senders list,
+        this method has no effect."""
+        return self._native_obj.removeTrack(sender._native_obj)
+
+    def restart_ice(self) -> None:
+        """Allows to easily request that ICE candidate gathering be redone on both ends of the connection.
+        This simplifies the process by allowing the same method to be used by either the caller or the receiver
+        to trigger an ICE restart."""
+        return self._native_obj.restartIce()
+
     def close(self):
         """Closes the current peer connection."""
         return self._native_obj.close()
 
+    @property
+    def sctp(self) -> Optional['webrtc.RTCSctpTransport']:
+        """:obj:`webrtc.RTCSctpTransport`, optional: An object describing the SCTP transport layer over which SCTP
+        data is being sent and received. If SCTP hasn't been negotiated, this value is :obj:`None`"""
+        from webrtc import RTCSctpTransport
+
+        sctp = self._native_obj.sctp
+        if sctp:
+            return RTCSctpTransport._wrap(sctp)
+
+        return None
+
     #: Alias for :attr:`create_offer`
     createOffer = create_offer
     #: Alias for :attr:`create_answer`
@@ -207,3 +232,7 @@ def close(self):
     getSenders = get_senders
     #: Alias for :attr:`get_receivers`
     getReceivers = get_receivers
+    #: Alias for :attr:`remove_track`
+    removeTrack = remove_track
+    #: Alias for :attr:`restart_ice`
+    restartIce = restart_ice