#!/usr/bin/env python3

# Libervia WebRTC implementation
# Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org)

# This program is free software: you can redistribute it and/or modify
# it under the terms of the GNU Affero General Public License as published by
# the Free Software Foundation, either version 3 of the License, or
# (at your option) any later version.

# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
# GNU Affero General Public License for more details.

# You should have received a copy of the GNU Affero General Public License
# along with this program.  If not, see <http://www.gnu.org/licenses/>.

from collections.abc import Awaitable
import gi

gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"})
from gi.repository import Gst, GstWebRTC, GstSdp

from libervia.backend.core import exceptions

try:
    from gi.overrides import Gst as _
except ImportError:
    raise exceptions.MissingModule(
        "No GStreamer Python overrides available. Please install relevant packages on "
        "your system (e.g., `python3-gst-1.0` on Debian and derivatives)."
    )
import asyncio
from datetime import datetime
import logging
import re
from typing import Callable, Final
from urllib.parse import quote_plus

from libervia.backend.tools.common import data_format
from libervia.frontends.tools import aio, display_servers, jid
from .webrtc_models import (
    CallData,
    SinksApp,
    SinksAuto,
    SinksData,
    SinksDataChannel,
    SinksNone,
    SourcesAuto,
    SourcesData,
    SourcesDataChannel,
    SourcesNone,
    SourcesPipeline,
    SourcesTest,
)

current_server = display_servers.detect()
if current_server == display_servers.X11:
    # GSTreamer's ximagesrc documentation asks to run this function
    import ctypes

    ctypes.CDLL("libX11.so.6").XInitThreads()


log = logging.getLogger(__name__)
VIDEO_SOURCE_AUTO: Final = "v4l2src"
AUDIO_SOURCE_AUTO: Final = "pulsesrc"
NONE_NOT_IMPLEMENTED_MSG: Final = "None value is not handled yet."

Gst.init(None)


class WebRTC:
    """GSTreamer based WebRTC implementation for audio and video communication.

    This class encapsulates the WebRTC functionalities required for initiating and
    handling audio and video calls, and data channels.
    """

    def __init__(
        self,
        bridge,
        profile: str,
        sources_data: SourcesData|None = None,
        sinks_data: SinksData|None = None,
        reset_cb: Callable | None = None,
        merge_pip: bool | None = None,
        target_size: tuple[int, int] | None = None,
        call_start_cb: Callable[[str, dict, str], Awaitable[str]] | None = None,
        dc_data_list: list[SourcesDataChannel|SinksDataChannel]|None = None
    ) -> None:
        """Initializes a new WebRTC instance.

        @param bridge: An instance of backend bridge.
        @param profile: Libervia profile.
        @param sources_data: Data of the sources.
            The model used will determine which sources to use.
            SourcesDataChannel can be used here as a convenience. It will then be moved
            to ``data_channels`` and ``SourcesNone`` will be used instead for
            ``sources_data``.
            If None, SourcesAuto will be used.
        @param sinks_data: Data of the sinks.
            The model used will determine which sinks to use.
            SinksDataChannel can be used here as a convenience. It will then be moved
            to ``data_channels`` and ``SinksNone`` will be used instead for
            ``sinks_data``.
            If None, SinksAuto will be used.
        @param reset_cb: An optional Callable that is triggered on reset events. Can be
            used to reset UI data on new calls.
        @param merge_pip: A boolean flag indicating whether Picture-in-Picture mode is
            enabled. When PiP is used, local feedback is merged to remote video stream.
            Only one video stream is then produced (the local one).
            If None, PiP mode is selected automatically according to selected sink (it's
            used for SinksAuto only for now).
        @param target_size: Expected size of the final sink stream. Mainly use by composer
            when ``merge_pip`` is set.
            None to autodetect (no real autodetection implemeted yet, default to
            (1280,720)).
        @param call_start_cb: Called when call is started.
        @param dc_data_list: Data Channels to create.
            If a SourcesDataChannel is used as ``sources_data``, or a SinksDataChannel is
            used as ``sinks_data``, they will be automatically added to this list.
        """
        self.main_loop = asyncio.get_event_loop()
        self.bridge = bridge
        self.profile = profile
        self.pipeline = None
        self._audio_muted = False
        self._video_muted = False
        self._desktop_sharing = False
        self.desktop_sharing_data = None
        if dc_data_list is None:
            dc_data_list = []
        self.dc_data_list = dc_data_list
        if sources_data is None:
            sources_data = SourcesAuto()
        elif isinstance(sources_data, SourcesDataChannel):
            dc_data_list.append(sources_data)
            sources_data = SourcesNone()
        self.sources_data = sources_data
        if sinks_data is None:
            sinks_data = SinksAuto()
        elif isinstance(sinks_data, SinksDataChannel):
            dc_data_list.append(sinks_data)
            sinks_data = SinksNone()
        self.sinks_data = sinks_data
        if target_size is None:
            target_size = (1280, 720)
        self.target_width, self.target_height = target_size
        if merge_pip is None:
            merge_pip = isinstance(sinks_data, SinksAuto)
        self.merge_pip = merge_pip
        if call_start_cb is None:
            call_start_cb = self._call_start
        self.call_start_cb = call_start_cb
        if isinstance(sinks_data, SinksApp):
            if merge_pip and sinks_data.remote_video_cb is not None:
                raise ValueError("Remote_video_cb can't be used when merge_pip is used!")
        self.reset_cb = reset_cb
        if current_server == display_servers.WAYLAND:
            from .portal_desktop import DesktopPortal

            self.desktop_portal = DesktopPortal(
                on_session_closed_cb=self.on_portal_session_closed
            )
        else:
            self.desktop_portal = None
        self.reset_instance()

    @property
    def audio_muted(self):
        return self._audio_muted

    @audio_muted.setter
    def audio_muted(self, muted: bool) -> None:
        if muted != self._audio_muted:
            self._audio_muted = muted
            self.on_audio_mute(muted)

    @property
    def video_muted(self):
        return self._video_muted

    @video_muted.setter
    def video_muted(self, muted: bool) -> None:
        if muted != self._video_muted:
            self._video_muted = muted
            self.on_video_mute(muted)

    @property
    def desktop_sharing(self):
        return self._desktop_sharing

    @desktop_sharing.setter
    def desktop_sharing(self, active: bool) -> None:
        if active != self._desktop_sharing:
            self._desktop_sharing = active
            self.on_desktop_switch(active)
            try:
                cb = self.bindings["desktop_sharing"]
            except KeyError:
                pass
            else:
                cb(active)

    @property
    def sdp_set(self):
        return self._sdp_set

    @sdp_set.setter
    def sdp_set(self, is_set: bool):
        self._sdp_set = is_set
        if is_set:
            self.on_ice_candidates_new(self.remote_candidates_buffer)
            for data in self.remote_candidates_buffer.values():
                data["candidates"].clear()

    @property
    def media_types(self):
        if self._media_types is None:
            raise Exception("self._media_types should not be None!")
        return self._media_types

    @media_types.setter
    def media_types(self, new_media_types: dict) -> None:
        self._media_types = new_media_types
        self._media_types_inv = {v: k for k, v in new_media_types.items()}

    @property
    def media_types_inv(self) -> dict:
        if self._media_types_inv is None:
            raise Exception("self._media_types_inv should not be None!")
        return self._media_types_inv

    def bind(self, **kwargs: Callable) -> None:
        self.bindings.clear()
        for key, cb in kwargs.items():
            if key not in ("desktop_sharing",):
                raise ValueError(
                    'Only "desktop_sharing" is currently allowed for binding'
                )
            self.bindings[key] = cb

    def generate_dot_file(
        self,
        filename: str = "pipeline",
        details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL,
        with_timestamp: bool = True,
        bin_: Gst.Bin | None = None,
    ) -> None:
        """Generate Dot File for debugging

        ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir.
        ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file.
        See
        https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails
        for details.

        @param filename: name of the generated file
        @param details: which details to print
        @param with_timestamp: if True, add a timestamp to filename
        @param bin_: which bin to output. By default, the whole pipeline
            (``self.pipeline``) will be used.
        """
        if bin_ is None:
            bin_ = self.pipeline
        if with_timestamp:
            timestamp = datetime.now().isoformat(timespec="milliseconds")
            filename = f"{timestamp}_filename"

        Gst.debug_bin_to_dot_file(bin_, details, filename)

    def get_sdp_mline_index(self, media_type: str) -> int:
        """Gets the sdpMLineIndex for a given media type.

        @param media_type: The type of the media.
        """
        for index, m_type in self.media_types.items():
            if m_type == media_type:
                return index
        raise ValueError(f"Media type '{media_type}' not found")

    def _set_media_types(self, offer_sdp: str) -> None:
        """Sets media types from offer SDP

        @param offer: RTC session description containing the offer
        """
        sdp_lines = offer_sdp.splitlines()
        media_types = {}
        mline_index = 0

        for line in sdp_lines:
            if line.startswith("m="):
                media_types[mline_index] = line[2 : line.find(" ")]
                mline_index += 1

        self.media_types = media_types

    def _a_call(self, method, *args, **kwargs):
        """Call an async method in main thread"""
        aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop)

    def get_payload_types(
        self, sdpmsg, video_encoding: str, audio_encoding: str
    ) -> dict[str, int | None]:
        """Find the payload types for the specified video and audio encoding.

        Very simplistically finds the first payload type matching the encoding
        name. More complex applications will want to match caps on
        profile-level-id, packetization-mode, etc.
        """
        # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at
        # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
        video_pt = None
        audio_pt = None
        for i in range(0, sdpmsg.medias_len()):
            media = sdpmsg.get_media(i)
            for j in range(0, media.formats_len()):
                fmt = media.get_format(j)
                if fmt == "webrtc-datachannel":
                    continue
                pt = int(fmt)
                caps = media.get_caps_from_media(pt)
                s = caps.get_structure(0)
                encoding_name = s["encoding-name"]
                if video_pt is None and encoding_name == video_encoding:
                    video_pt = pt
                elif audio_pt is None and encoding_name == audio_encoding:
                    audio_pt = pt
        return {video_encoding: video_pt, audio_encoding: audio_pt}

    def parse_ice_candidate(self, candidate_string):
        """Parses the ice candidate string.

        @param candidate_string: The ice candidate string to be parsed.
        """
        pattern = re.compile(
            r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) "
            r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ "
            r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport "
            r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?"
        )
        match = pattern.match(candidate_string)
        if match:
            candidate_dict = match.groupdict()

            # Apply the correct types to the dictionary values
            candidate_dict["component_id"] = int(candidate_dict["component_id"])
            candidate_dict["priority"] = int(candidate_dict["priority"])
            candidate_dict["port"] = int(candidate_dict["port"])

            if candidate_dict["rel_port"]:
                candidate_dict["rel_port"] = int(candidate_dict["rel_port"])

            if candidate_dict["generation"]:
                candidate_dict["generation"] = candidate_dict["generation"]

            # Remove None values
            return {k: v for k, v in candidate_dict.items() if v is not None}
        else:
            log.warning(f"can't parse candidate: {candidate_string!r}")
            return None

    def build_ice_candidate(self, parsed_candidate):
        """Builds ICE candidate

        @param parsed_candidate: Dictionary containing parsed ICE candidate
        """
        base_format = (
            "candidate:{foundation} {component_id} {transport} {priority} "
            "{address} {port} typ {type}"
        )

        if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"):
            base_format += " raddr {rel_addr} rport {rel_port}"

        if parsed_candidate.get("generation"):
            base_format += " generation {generation}"

        return base_format.format(**parsed_candidate)

    def extract_ufrag_pwd(self, sdp: str) -> None:
        """Retrieves ICE password and user fragment for SDP offer.

        @param sdp: The Session Description Protocol offer string.
        """
        lines = sdp.splitlines()
        media = ''
        mid_media_map = {}
        bundle_media = set()
        bundle_ufrag = ''
        bundle_pwd = ''
        in_bundle = False

        for line in lines:
            if line.startswith('m='):
                media = line.split('=')[1].split()[0]
            elif line.startswith('a=mid:'):
                mid = line.split(':')[1].strip()
                mid_media_map[mid] = media
            elif line.startswith('a=group:BUNDLE'):
                in_bundle = True
                bundle_media = set(line.split(':')[1].strip().split())
            elif line.startswith('a=ice-ufrag:'):
                if in_bundle:
                    bundle_ufrag = line.split(':')[1].strip()
                else:
                    self.ufrag[media] = line.split(':')[1].strip()
            elif line.startswith('a=ice-pwd:'):
                if in_bundle:
                    bundle_pwd = line.split(':')[1].strip()
                else:
                    self.pwd[media] = line.split(':')[1].strip()
            else:
                in_bundle = False

        if bundle_ufrag and bundle_pwd:
            for mid in bundle_media:
                media = mid_media_map[mid]
                self.ufrag[media] = bundle_ufrag
                self.pwd[media] = bundle_pwd

    def reset_instance(self):
        """Inits or resets the instance variables to their default state."""
        self.role: str | None = None
        if self.pipeline is not None:
            self.pipeline.set_state(Gst.State.NULL)
        self.pipeline = None
        self._remote_video_pad = None
        self.sid: str | None = None
        self.offer: str | None = None
        self.local_candidates_buffer = {}
        self.ufrag: dict[str, str] = {}
        self.pwd: dict[str, str] = {}
        self.callee: jid.JID | None = None
        self._media_types = None
        self._media_types_inv = None
        self._sdp_set: bool = False
        self.remote_candidates_buffer: dict[str, dict[str, list]] = {
            "audio": {"candidates": []},
            "video": {"candidates": []},
        }
        self._media_types = None
        self._media_types_inv = None
        self.audio_valve = None
        self.video_valve = None
        self.video_selector = None
        if self.desktop_portal is not None:
            self.desktop_portal.end_session()
        self.desktop_sharing = False
        self.desktop_sink_pad = None
        self.bindings = {}
        if self.reset_cb is not None:
            self.reset_cb()
        self.data_channels: dict[str, GstWebRTC.WebRTCDataChannel] = {}

    @property
    def data_channel(self) -> GstWebRTC.WebRTCDataChannel:
        """Convenience method to get WebRTCDataChannel instance when there is only one."""
        if len(self.data_channels) != 1:
            raise exceptions.InternalError(
                "self.data_channel can only be used in a single Data Channel scenario. "
                "Use self.data_channels dict instead."
            )
        return next(iter(self.data_channels.values()))

    async def setup_call(
        self,
        role: str,
        audio_pt: int | None = 96,
        video_pt: int | None = 97,
    ) -> None:
        """Sets up the call.

        This method establishes the Gstreamer pipeline for audio and video communication.
        The method also manages STUN and TURN server configurations, signal watchers, and
        various connection handlers for the webrtcbin.

        @param role: The role for the call, either 'initiator' or 'responder'.
        @param audio_pt: The payload type for the audio stream.
        @param video_pt: The payload type for the video stream

        @raises NotImplementedError: If audio_pt or video_pt is set to None.
        @raises AssertionError: If the role is not 'initiator' or 'responder'.
        """
        assert role in ("initiator", "responder")
        self.role = role


        if isinstance(self.sources_data, SourcesPipeline):
            if self.sources_data.video_pipeline!= "" and video_pt is None:
                raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG)
            if self.sources_data.audio_pipeline!= "" and audio_pt is None:
                raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG)
        elif isinstance(self.sources_data, SourcesNone):
            pass
        else:
            if audio_pt is None or video_pt is None:
                raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG)

        match self.sources_data:
            case SourcesAuto():
                video_source_elt = VIDEO_SOURCE_AUTO
                audio_source_elt = AUDIO_SOURCE_AUTO
            case SourcesNone():
                video_source_elt = ""
                audio_source_elt = ""
            case SourcesPipeline() as source:
                if source.video_pipeline is None:
                    video_source_elt = VIDEO_SOURCE_AUTO
                else:
                    video_source_elt = source.video_pipeline
                if source.audio_pipeline is None:
                    audio_source_elt = AUDIO_SOURCE_AUTO
                else:
                    audio_source_elt = source.audio_pipeline
            case SourcesTest():
                video_source_elt = "videotestsrc is-live=true pattern=ball"
                audio_source_elt = "audiotestsrc"
            case _:
                raise exceptions.InternalError(
                    f'Unexpected "sources_data" value: {self.sources_data!r}'
                )

        match self.sinks_data:
            case SinksApp():
                local_video_sink_elt = (
                    "appsink name=local_video_sink emit-signals=true drop=true "
                    "max-buffers=1 sync=True"
                )
            case SinksAuto():
                if isinstance(self.sources_data, SourcesNone):
                    local_video_sink_elt = ""
                else:
                    local_video_sink_elt = "autovideosink"
            case SinksNone():
                local_video_sink_elt = ""
            case _:
                raise exceptions.InternalError(
                    f'Unexpected "sinks_data" value {self.sinks_data!r}'
                )

        gst_pipe_elements = [
            "webrtcbin latency=30 name=sendrecv bundle-policy=max-bundle"
        ]

        if self.merge_pip and local_video_sink_elt:
            # Compositor is used to merge local video feedback in video sink, useful when
            # we have only a single video sink.
            gst_pipe_elements.append(
                "compositor name=compositor background=black "
                f"! video/x-raw,width={self.target_width},"
                f"height={self.target_height},framerate=30/1 "
                f"! {local_video_sink_elt}"
            )
            local_video_sink_elt = "compositor.sink_1"

        if video_source_elt:
            # Video source with an input-selector to switch between normal and video mute
            # (or desktop sharing).
            gst_pipe_elements.append(f"""
        input-selector name=video_selector
        ! videorate drop-only=1 max-rate=30
        ! video/x-raw,framerate=30/1
        ! tee name=t

        {video_source_elt} name=video_src
        ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! video_selector.

        videotestsrc name=muted_src is-live=true pattern=black
        ! queue leaky=downstream
        ! video_selector.

        t.
        ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! videoscale
        ! videoconvert
        ! vp8enc deadline=1 keyframe-max-dist=30
        ! rtpvp8pay picture-id-mode=15-bit
        ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt}
        ! sendrecv.
        """)

        if local_video_sink_elt:
            # Local video feedback.
            gst_pipe_elements.append(f"""
        t.
        ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! videoconvert
        ! {local_video_sink_elt}
        """)

        if audio_source_elt:
            # Audio with a valve for muting.
            gst_pipe_elements.append(r"""
        {audio_source_elt} name=audio_src
        ! valve
        ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! audioconvert
        ! audioresample
        ! opusenc audio-type=voice
        ! rtpopuspay
        ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt}
        ! sendrecv.
        """)

        self.gst_pipe_desc = "\n\n".join(gst_pipe_elements)

        log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}")

        # Create the pipeline
        try:
            self.pipeline = Gst.parse_launch(self.gst_pipe_desc)
        except Exception:
            log.exception("Can't parse pipeline")
            self.pipeline = None
        if not self.pipeline:
            raise exceptions.InternalError("Failed to create Gstreamer pipeline.")

        if not isinstance(self.pipeline, Gst.Pipeline):
            # in the case of Data Channel there is a single element, and Gst.parse_launch
            # doesn't create a Pipeline in this case, so we do it manually.
            pipeline = Gst.Pipeline()
            pipeline.add(self.pipeline)
            self.pipeline = pipeline

        self.webrtcbin = self.pipeline.get_by_name("sendrecv")
        if self.webrtcbin is None:
            raise exceptions.InternalError("Can't get the pipeline.")

        # If video or audio sources are not created, ``get_by_name`` will return None.
        self.video_src = self.pipeline.get_by_name("video_src")
        self.muted_src = self.pipeline.get_by_name("muted_src")
        self.video_selector = self.pipeline.get_by_name("video_selector")
        if self.video_src and isinstance(self.sources_data, SourcesPipeline):
            for name, value in self.sources_data.video_properties.items():
                self.video_src.set_property(name, value)

        self.audio_src = self.pipeline.get_by_name("audio_src")
        if self.audio_src and isinstance(self.sources_data, SourcesPipeline):
            for name, value in self.sources_data.audio_properties.items():
                self.audio_src.set_property(name, value)

        self.audio_valve = self.pipeline.get_by_name("audio_valve")

        if self.video_muted:
            self.on_video_mute(True)
        if self.audio_muted:
            self.on_audio_mute(True)

        # set STUN and TURN servers
        external_disco = data_format.deserialise(
            await self.bridge.external_disco_get("", self.profile), type_check=list
        )

        for server in external_disco:
            if server["type"] == "stun":
                if server["transport"] == "tcp":
                    log.info(
                        "ignoring TCP STUN server, GStreamer only support one STUN server"
                    )
                url = f"stun://{server['host']}:{server['port']}"
                log.debug(f"adding stun server: {url}")
                self.webrtcbin.set_property("stun-server", url)
            elif server["type"] == "turn":
                url = "{scheme}://{username}:{password}@{host}:{port}".format(
                    scheme="turns" if server["transport"] == "tcp" else "turn",
                    username=quote_plus(server["username"]),
                    password=quote_plus(server["password"]),
                    host=server["host"],
                    port=server["port"],
                )
                log.debug(f"adding turn server: {url}")

                if not self.webrtcbin.emit("add-turn-server", url):
                    log.warning(f"Erreur while adding TURN server {url}")

        # local video feedback
        if isinstance(self.sinks_data, SinksApp):
            local_video_sink = self.pipeline.get_by_name("local_video_sink")
            if local_video_sink is not None:
                local_video_sink.set_property("emit-signals", True)
                local_video_sink.connect("new-sample", self.sinks_data.local_video_cb)
                local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB")
                local_video_sink.set_property("caps", local_video_sink_caps)

        # Create bus and associate signal watchers
        self.bus = self.pipeline.get_bus()
        if not self.bus:
            log.error("Failed to get bus from pipeline.")
            return

        self.bus.add_signal_watch()
        self.webrtcbin.connect("pad-added", self.on_pad_added)
        self.bus.connect("message::error", self.on_bus_error)
        self.bus.connect("message::eos", self.on_bus_eos)
        self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed)
        self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate)
        self.webrtcbin.connect(
            "notify::ice-gathering-state", self.on_ice_gathering_state_change
        )
        self.webrtcbin.connect(
            "notify::ice-connection-state", self.on_ice_connection_state
        )

        for dc_data in self.dc_data_list:
            self.create_data_channel(dc_data)

    def create_data_channel(self, dc_data: SourcesDataChannel|SinksDataChannel) -> None:
        """Create a Data Channel and connect relevant callbacks."""
        assert self.pipeline is not None
        if isinstance(dc_data, SourcesDataChannel):
            # Data channel configuration for compatibility with browser defaults
            data_channel_options = Gst.Structure.new_empty("data-channel-options")
            data_channel_options.set_value("ordered", True)
            data_channel_options.set_value("protocol", "")

            # Create the data channel
            self.pipeline.set_state(Gst.State.READY)
            self.data_channels[dc_data.name] = data_channel = self.webrtcbin.emit(
                "create-data-channel", dc_data.name, data_channel_options
            )
            if data_channel is None:
                log.error("Failed to create data channel")
                return
            data_channel.connect("on-open", dc_data.dc_open_cb)
        elif isinstance(dc_data, SinksDataChannel):
            self.webrtcbin.connect("on-data-channel", dc_data.dc_on_data_channel)
        else:
            raise ValueError(
                "Only SourcesDataChannel or SinksDataChannel are allowed."
            )

    def start_pipeline(self) -> None:
        """Starts the GStreamer pipeline."""
        log.debug("starting the pipeline")
        self.pipeline.set_state(Gst.State.PLAYING)

    def on_negotiation_needed(self, webrtc):
        """Initiate SDP offer when negotiation is needed."""
        log.debug("Negotiation needed.")
        if self.role == "initiator":
            log.debug("Creating offer…")
            promise = Gst.Promise.new_with_change_func(self.on_offer_created)
            self.webrtcbin.emit("create-offer", None, promise)

    def on_offer_created(self, promise):
        """Callback for when SDP offer is created."""
        log.info("on_offer_created called")
        assert promise.wait() == Gst.PromiseResult.REPLIED
        reply = promise.get_reply()
        if reply is None:
            log.error("Promise reply is None. Offer creation might have failed.")
            return
        offer = reply["offer"]
        self.offer = offer.sdp.as_text()
        log.info(f"SDP offer created: \n{self.offer}")
        self._set_media_types(self.offer)
        promise = Gst.Promise.new()
        self.webrtcbin.emit("set-local-description", offer, promise)
        promise.interrupt()
        self._a_call(self._start_call)

    def on_answer_set(self, promise):
        assert promise.wait() == Gst.PromiseResult.REPLIED

    def on_answer_created(self, promise, _, __):
        """Callback for when SDP answer is created."""
        assert promise.wait() == Gst.PromiseResult.REPLIED
        reply = promise.get_reply()
        answer = reply["answer"]
        promise = Gst.Promise.new()
        self.webrtcbin.emit("set-local-description", answer, promise)
        promise.interrupt()
        answer_sdp = answer.sdp.as_text()
        log.info(f"SDP answer set: \n{answer_sdp}")
        self.sdp_set = True
        self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile)

    def on_offer_set(self, promise):
        assert promise.wait() == Gst.PromiseResult.REPLIED
        promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
        self.webrtcbin.emit("create-answer", None, promise)

    def link_element_or_pad(
        self, source: Gst.Element, dest: Gst.Element | Gst.Pad
    ) -> bool:
        """Check if dest is a pad or an element, and link appropriately"""
        src_pad = source.get_static_pad("src")

        if isinstance(dest, Gst.Pad):
            # If the dest is a pad, link directly
            if not src_pad.link(dest) == Gst.PadLinkReturn.OK:
                log.error(
                    "Failed to link 'conv' to the compositor's newly requested pad!"
                )
                return False
        elif isinstance(dest, Gst.Element):
            return source.link(dest)
        else:
            log.error(f"Unexpected type for dest: {type(dest)}")
            return False

        return True

    def scaled_dimensions(
        self, original_width: int, original_height: int, max_width: int, max_height: int
    ) -> tuple[int, int]:
        """Calculates the scaled dimensions preserving aspect ratio.

        @param original_width: Original width of the video stream.
        @param original_height: Original height of the video stream.
        @param max_width: Maximum desired width for the scaled video.
        @param max_height: Maximum desired height for the scaled video.
        @return: The width and height of the scaled video.
        """
        aspect_ratio = original_width / original_height
        new_width = int(max_height * aspect_ratio)

        if new_width <= max_width:
            return new_width, max_height

        new_height = int(max_width / aspect_ratio)
        return max_width, new_height

    def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None:
        """Handle the stream from the remote decodebin.

        This method processes the incoming stream from the remote decodebin, determining
        whether it's video or audio. It then sets up the appropriate GStreamer elements
        for video/audio processing and adds them to the pipeline.

        @param pad: The Gst.Pad from the remote decodebin producing the stream.
        """
        assert self.pipeline is not None
        if not pad.has_current_caps():
            log.error(f"{pad} has no caps, ignoring")
            return

        caps = pad.get_current_caps()
        assert len(caps)
        s = caps[0]
        name = s.get_name()
        log.debug(f"====> NAME START: {name}")

        q = Gst.ElementFactory.make("queue")

        if name.startswith("video"):
            log.debug("===> VIDEO OK")

            self._remote_video_pad = pad

            # Check and log the original size of the video
            width = self.target_width
            height = self.target_height
            log.info(f"Original video size: {width}x{height}")

            # This is a fix for an issue found with Movim on desktop: a non standard
            # resolution is used (990x557) resulting in bad alignement and no color in
            # rendered image
            adjust_resolution = width % 4 != 0 or height % 4 != 0
            if adjust_resolution:
                log.info("non standard resolution, we need to adjust size")
                width = (width + 3) // 4 * 4
                height = (height + 3) // 4 * 4
                log.info(f"Adjusted video size: {width}x{height}")

            conv = Gst.ElementFactory.make("videoconvert")
            if self.merge_pip:
                # with ``merge_pip`` set, we plug the remote stream to the composer
                compositor = self.pipeline.get_by_name("compositor")

                sink1_pad = compositor.get_static_pad("sink_1")

                local_width, local_height = self.scaled_dimensions(
                    sink1_pad.get_property("width"),
                    sink1_pad.get_property("height"),
                    width // 3,
                    height // 3,
                )

                sink1_pad.set_property("xpos", width - local_width)
                sink1_pad.set_property("ypos", height - local_height)
                sink1_pad.set_property("width", local_width)
                sink1_pad.set_property("height", local_height)
                sink1_pad.set_property("sizing-policy", 1)
                sink1_pad.set_property("zorder", 1)

                # Request a new pad for the remote stream
                sink_pad_template = compositor.get_pad_template("sink_%u")
                remote_video_sink = compositor.request_pad(sink_pad_template, None, None)
                remote_video_sink.set_property("zorder", 0)
                remote_video_sink.set_property("width", width)
                remote_video_sink.set_property("height", height)
                remote_video_sink.set_property("sizing-policy", 1)
            elif isinstance(self.sinks_data, SinksApp):
                # ``app`` sink without ``self.merge_pip`` set, be create the sink and
                # connect it to the ``remote_video_cb``.
                remote_video_sink = Gst.ElementFactory.make("appsink")

                remote_video_caps = Gst.Caps.from_string("video/x-raw,format=RGB")
                remote_video_sink.set_property("caps", remote_video_caps)

                remote_video_sink.set_property("emit-signals", True)
                remote_video_sink.set_property("drop", True)
                remote_video_sink.set_property("max-buffers", 1)
                remote_video_sink.set_property("sync", True)
                remote_video_sink.connect("new-sample", self.sinks_data.remote_video_cb)
                self.pipeline.add(remote_video_sink)
            elif isinstance(self.sinks_data, SinksAuto):
                # if ``self.merge_pip`` is not set, we create a dedicated
                # ``autovideosink`` for remote stream.
                remote_video_sink = Gst.ElementFactory.make("autovideosink")
                self.pipeline.add(remote_video_sink)
            else:
                raise exceptions.InternalError(
                    f'Unhandled "sinks_data" value: {self.sinks_data!r}'
                )

            if adjust_resolution:
                videoscale = Gst.ElementFactory.make("videoscale")
                adjusted_caps = Gst.Caps.from_string(
                    f"video/x-raw,width={width},height={height}"
                )
                capsfilter = Gst.ElementFactory.make("capsfilter")
                capsfilter.set_property("caps", adjusted_caps)

                self.pipeline.add(q, conv, videoscale, capsfilter)

                self.pipeline.sync_children_states()
                ret = pad.link(q.get_static_pad("sink"))
                if ret != Gst.PadLinkReturn.OK:
                    log.error(f"Error linking pad: {ret}")
                q.link(conv)
                conv.link(videoscale)
                videoscale.link(capsfilter)
                self.link_element_or_pad(capsfilter.link, remote_video_sink)

            else:
                self.pipeline.add(q, conv)

                self.pipeline.sync_children_states()
                ret = pad.link(q.get_static_pad("sink"))
                if ret != Gst.PadLinkReturn.OK:
                    log.error(f"Error linking pad: {ret}")
                q.link(conv)
                self.link_element_or_pad(conv, remote_video_sink)

        elif name.startswith("audio"):
            log.debug("===> Audio OK")
            conv = Gst.ElementFactory.make("audioconvert")
            resample = Gst.ElementFactory.make("audioresample")
            remote_audio_sink = Gst.ElementFactory.make("autoaudiosink")
            self.pipeline.add(q, conv, resample, remote_audio_sink)
            self.pipeline.sync_children_states()
            ret = pad.link(q.get_static_pad("sink"))
            if ret != Gst.PadLinkReturn.OK:
                log.error(f"Error linking pad: {ret}")
            q.link(conv)
            conv.link(resample)
            resample.link(remote_audio_sink)

        else:
            log.warning(f"unmanaged name: {name!r}")

    def on_pad_added(self, __, pad: Gst.Pad) -> None:
        """Handle the addition of a new pad to the element.

        When a new source pad is added to the element, this method creates a decodebin,
        connects it to handle the stream, and links the pad to the decodebin.

        @param __: Placeholder for the signal source. Not used in this method.
        @param pad: The newly added pad.
        """
        log.debug("on_pad_added")
        if pad.direction != Gst.PadDirection.SRC:
            return

        decodebin = Gst.ElementFactory.make("decodebin")
        decodebin.connect("pad-added", self.on_remote_decodebin_stream)
        self.pipeline.add(decodebin)
        decodebin.sync_state_with_parent()
        pad.link(decodebin.get_static_pad("sink"))

    async def _call_start(self, callee: jid.JID, call_data: dict, profile: str) -> str:
        return await self.bridge.call_start(
            str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile
        )

    async def _start_call(self) -> None:
        """Initiate the call.

        Initiates a call with the callee using the stored offer. If there are any buffered
        local ICE candidates, they are sent as part of the initiation.
        """
        assert self.callee
        assert self.call_start_cb is not None
        self.sid = await self.call_start_cb(
            self.callee, {"sdp": self.offer}, self.profile
        )
        if self.local_candidates_buffer:
            log.debug(
                f"sending buffered local ICE candidates: {self.local_candidates_buffer}"
            )
            if not self.pwd:
                sdp = self.webrtcbin.props.local_description.sdp.as_text()
                self.extract_ufrag_pwd(sdp)
            ice_data = {}
            for media_type, candidates in self.local_candidates_buffer.items():
                ice_data[media_type] = {
                    "ufrag": self.ufrag[media_type],
                    "pwd": self.pwd[media_type],
                    "candidates": candidates,
                }
            await self.bridge.ice_candidates_add(
                self.sid, data_format.serialise(ice_data), self.profile
            )
            self.local_candidates_buffer.clear()

    def _remote_sdp_set(self, promise) -> None:
        assert promise.wait() == Gst.PromiseResult.REPLIED
        self.sdp_set = True

    def on_accepted_call(self, sdp: str, profile: str) -> None:
        """Outgoing call has been accepted.

        @param sdp: The SDP answer string received from the other party.
        @param profile: Profile used for the call.
        """
        log.debug(f"SDP answer received: \n{sdp}")

        __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
        answer = GstWebRTC.WebRTCSessionDescription.new(
            GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg
        )
        promise = Gst.Promise.new_with_change_func(self._remote_sdp_set)
        self.webrtcbin.emit("set-remote-description", answer, promise)

    async def answer_call(self, sdp: str, profile: str) -> None:
        """Answer an incoming call

        @param sdp: The SDP offer string received from the initiator.
        @param profile: Profile used for the call.

        @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in
            payload types.
        """
        log.debug(f"SDP offer received: \n{sdp}")
        self._set_media_types(sdp)
        __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp)
        payload_types = self.get_payload_types(
            offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS"
        )
        assert "VP8" in payload_types
        assert "OPUS" in payload_types
        try:
            await self.setup_call(
                "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"]
            )
        except Exception:
            log.exception("Can't setup call")
            raise
        self.start_pipeline()
        offer = GstWebRTC.WebRTCSessionDescription.new(
            GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg
        )
        promise = Gst.Promise.new_with_change_func(self.on_offer_set)
        self.webrtcbin.emit("set-remote-description", offer, promise)

    def on_ice_candidate(self, webrtc, mline_index, candidate_sdp):
        """Handles the on-ice-candidate signal of webrtcbin.

        @param webrtc: The webrtcbin element.
        @param mlineindex: The mline index.
        @param candidate: The ICE candidate.
        """
        log.debug(
            f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}"
        )
        parsed_candidate = self.parse_ice_candidate(candidate_sdp)
        if parsed_candidate is None:
            log.warning(f"Can't parse candidate: {candidate_sdp}")
            return
        try:
            media_type = self.media_types[mline_index]
        except KeyError:
            raise exceptions.InternalError("can't find media type")

        if self.sid is None:
            log.debug("buffering local ICE candidate")
            self.local_candidates_buffer.setdefault(media_type, []).append(
                parsed_candidate
            )
        else:
            sdp = self.webrtcbin.props.local_description.sdp.as_text()
            assert sdp is not None
            self.extract_ufrag_pwd(sdp)
            ice_data = {
                "ufrag": self.ufrag[media_type],
                "pwd": self.pwd[media_type],
                "candidates": [parsed_candidate]
            }
            self._a_call(
                self.bridge.ice_candidates_add,
                self.sid,
                data_format.serialise({media_type: ice_data}),
                self.profile,
            )

    def on_ice_candidates_new(self, candidates: dict) -> None:
        """Handle new ICE candidates.

        @param candidates: A dictionary containing media types ("audio" or "video") as
            keys and corresponding ICE data as values.

        @raise exceptions.InternalError: Raised when sdp mline index is not found.
        """
        if not self.sdp_set:
            log.debug("buffering remote ICE candidate")
            for media_type in ("audio", "video"):
                media_candidates = candidates.get(media_type)
                if media_candidates:
                    buffer = self.remote_candidates_buffer[media_type]
                    buffer["candidates"].extend(media_candidates["candidates"])
            return
        for media_type, ice_data in candidates.items():
            for candidate in ice_data["candidates"]:
                candidate_sdp = self.build_ice_candidate(candidate)
                try:
                    mline_index = self.get_sdp_mline_index(media_type)
                except Exception as e:
                    raise exceptions.InternalError(f"Can't find sdp mline index: {e}")
                self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp)
                log.warning(
                    f"Remote ICE candidate added. MLine Index: {mline_index}, "
                    f"{candidate_sdp}"
                )

    def on_ice_gathering_state_change(self, pspec, __):
        state = self.webrtcbin.get_property("ice-gathering-state")
        log.debug(f"ICE gathering state changed to {state}")

    def on_ice_connection_state(self, pspec, __):
        state = self.webrtcbin.props.ice_connection_state
        if state == GstWebRTC.WebRTCICEConnectionState.FAILED:
            log.error("ICE connection failed")
        log.info(f"ICE connection state changed to {state}")

    def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None:
        """Handles the GStreamer bus error messages.

        @param bus: The GStreamer bus.
        @param message: The error message.
        """
        err, debug = message.parse_error()
        log.error(f"Error from {message.src.get_name()}: {err.message}")
        log.error(f"Debugging info: {debug}")

    def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None:
        """Handles the GStreamer bus eos messages.

        @param bus: The GStreamer bus.
        @param message: The eos message.
        """
        log.info("End of stream")

    def on_audio_mute(self, muted: bool) -> None:
        """Handles (un)muting of audio.

        @param muted: True if audio is muted.
        """
        if self.audio_valve is not None:
            self.audio_valve.set_property("drop", muted)
            state = "muted" if muted else "unmuted"
            log.info(f"audio is now {state}")

    def on_video_mute(self, muted: bool) -> None:
        """Handles (un)muting of video.

        @param muted: True if video is muted.
        """
        if self.video_selector is not None:
            current_source = (
                None if muted else "desktop" if self.desktop_sharing else "video"
            )
            self.switch_video_source(current_source)
            state = "muted" if muted else "unmuted"
            log.info(f"Video is now {state}")

    def on_desktop_switch(self, desktop_active: bool) -> None:
        """Switches the video source between desktop and video.

        @param desktop_active: True if desktop must be active. False for video.
        """
        if desktop_active and self.desktop_portal is not None:
            aio.run_async(self.on_desktop_switch_portal(desktop_active))
        else:
            self.do_desktop_switch(desktop_active)

    async def on_desktop_switch_portal(self, desktop_active: bool) -> None:
        """Call freedesktop screenshare portal and the activate the shared stream"""
        assert self.desktop_portal is not None
        try:
            screenshare_data = await self.desktop_portal.request_screenshare()
        except exceptions.CancelError:
            self.desktop_sharing = False
            return
        self.desktop_sharing_data = {"path": str(screenshare_data["node_id"])}
        self.do_desktop_switch(desktop_active)

    def on_portal_session_closed(self) -> None:
        self.desktop_sharing = False

    def do_desktop_switch(self, desktop_active: bool) -> None:
        if self.video_muted:
            # Update the active source state but do not switch
            self.desktop_sharing = desktop_active
            return

        source = "desktop" if desktop_active else "video"
        self.switch_video_source(source)
        self.desktop_sharing = desktop_active

    def switch_video_source(self, source: str | None) -> None:
        """Activates the specified source while deactivating the others.

        @param source: 'desktop', 'video', 'muted' or None for muted source.
        """
        if source is None:
            source = "muted"
        if source not in ["video", "muted", "desktop"]:
            raise ValueError(
                f"Invalid source: {source!r}, use one of {'video', 'muted', 'desktop'}"
            )

        self.pipeline.set_state(Gst.State.PAUSED)

        # Create a new desktop source if necessary
        if source == "desktop":
            self._setup_desktop_source(self.desktop_sharing_data)

        # Activate the chosen source and deactivate the others
        for src_name in ["video", "muted", "desktop"]:
            src_element = self.pipeline.get_by_name(f"{src_name}_src")
            if src_name == source:
                if src_element:
                    src_element.set_state(Gst.State.PLAYING)
            else:
                if src_element:
                    if src_name == "desktop":
                        self._remove_desktop_source(src_element)
                    else:
                        src_element.set_state(Gst.State.NULL)

        # Set the video_selector active pad
        if source == "desktop":
            if self.desktop_sink_pad:
                pad = self.desktop_sink_pad
            else:
                log.error(f"No desktop pad available")
                pad = None
        else:
            pad_name = f"sink_{['video', 'muted'].index(source)}"
            pad = self.video_selector.get_static_pad(pad_name)

        if pad is not None:
            self.video_selector.props.active_pad = pad

        self.pipeline.set_state(Gst.State.PLAYING)

    def _setup_desktop_source(self, properties: dict[str, object] | None) -> None:
        """Set up a new desktop source.

        @param properties: The properties to set on the desktop source.
        """
        source_elt = "ximagesrc" if self.desktop_portal is None else "pipewiresrc"
        desktop_src = Gst.ElementFactory.make(source_elt, "desktop_src")
        if properties is None:
            properties = {}
        for key, value in properties.items():
            log.debug(f"setting {source_elt} property: {key!r}={value!r}")
            desktop_src.set_property(key, value)
        video_convert = Gst.ElementFactory.make("videoconvert", "desktop_videoconvert")
        queue = Gst.ElementFactory.make("queue", "desktop_queue")
        queue.set_property("leaky", "downstream")

        self.pipeline.add(desktop_src)
        self.pipeline.add(video_convert)
        self.pipeline.add(queue)

        desktop_src.link(video_convert)
        video_convert.link(queue)

        sink_pad_template = self.video_selector.get_pad_template("sink_%u")
        self.desktop_sink_pad = self.video_selector.request_pad(
            sink_pad_template, None, None
        )
        queue_src_pad = queue.get_static_pad("src")
        queue_src_pad.link(self.desktop_sink_pad)

        desktop_src.sync_state_with_parent()
        video_convert.sync_state_with_parent()
        queue.sync_state_with_parent()

    def _remove_desktop_source(self, desktop_src: Gst.Element) -> None:
        """Remove the desktop source from the pipeline.

        @param desktop_src: The desktop source to remove.
        """
        # Remove elements for the desktop source
        video_convert = self.pipeline.get_by_name("desktop_videoconvert")
        queue = self.pipeline.get_by_name("desktop_queue")

        if video_convert:
            video_convert.set_state(Gst.State.NULL)
            desktop_src.unlink(video_convert)
            self.pipeline.remove(video_convert)

        if queue:
            queue.set_state(Gst.State.NULL)
            self.pipeline.remove(queue)

        desktop_src.set_state(Gst.State.NULL)
        self.pipeline.remove(desktop_src)

        # Release the pad associated with the desktop source
        if self.desktop_sink_pad:
            self.video_selector.release_request_pad(self.desktop_sink_pad)
            self.desktop_sink_pad = None

        if self.desktop_portal is not None:
            self.desktop_portal.end_session()

    async def end_call(self) -> None:
        """Stop streaming and clean instance"""
        self.reset_instance()


class WebRTCCall:
    """Helper class to create and handle WebRTC.

    This class handles signals and communication of connection data with backend.

    """

    def __init__(
        self,
        bridge,
        profile: str,
        callee: jid.JID,
        on_call_setup_cb: Callable | None = None,
        on_call_ended_cb: Callable | None = None,
        **kwargs,
    ):
        """Create and setup a webRTC instance

        @param bridge: async Bridge.
        @param profile: profile making or receiving the call
        @param callee: peer jid
        @param kwargs: extra kw args to use when instantiating WebRTC
        """
        self.profile = profile
        self.webrtc = WebRTC(bridge, profile, **kwargs)
        self.webrtc.callee = callee
        self.on_call_setup_cb = on_call_setup_cb
        self.on_call_ended_cb = on_call_ended_cb
        bridge.register_signal("ice_candidates_new", self.on_ice_candidates_new, "plugin")
        bridge.register_signal("call_setup", self.on_call_setup, "plugin")
        bridge.register_signal("call_ended", self.on_call_ended, "plugin")

    @classmethod
    async def make_webrtc_call(
        cls, bridge, profile: str, call_data: CallData, **kwargs
    ) -> "WebRTCCall":
        """Create the webrtc_call instance

        @param call_data: Call data of the command
        @param kwargs: extra args used to instanciate WebRTCCall

        """
        webrtc_call = cls(bridge, profile, call_data.callee, **call_data.kwargs, **kwargs)
        if call_data.sid is None:
            # we are making the call
            await webrtc_call.start()
        else:
            # we are receiving the call
            webrtc_call.sid = call_data.sid
            if call_data.action_id is not None:
                await bridge.action_launch(
                    call_data.action_id,
                    data_format.serialise({"cancelled": False}),
                    profile,
                )
        return webrtc_call

    @property
    def sid(self) -> str | None:
        return self.webrtc.sid

    @sid.setter
    def sid(self, new_sid: str | None) -> None:
        self.webrtc.sid = new_sid

    async def on_ice_candidates_new(
        self, sid: str, candidates_s: str, profile: str
    ) -> None:
        if sid != self.webrtc.sid or profile != self.profile:
            return
        self.webrtc.on_ice_candidates_new(
            data_format.deserialise(candidates_s),
        )

    async def on_call_setup(self, sid: str, setup_data_s: str, profile: str) -> None:
        if sid != self.webrtc.sid or profile != self.profile:
            return
        setup_data = data_format.deserialise(setup_data_s)
        try:
            role = setup_data["role"]
            sdp = setup_data["sdp"]
        except KeyError:
            log.error(f"Invalid setup data received: {setup_data}")
            return
        if role == "initiator":
            self.webrtc.on_accepted_call(sdp, profile)
        elif role == "responder":
            await self.webrtc.answer_call(sdp, profile)
        else:
            log.error(f"Invalid role received during setup: {setup_data}")
        if self.on_call_setup_cb is not None:
            await aio.maybe_async(self.on_call_setup_cb(sid, profile))

    async def on_call_ended(self, sid: str, data_s: str, profile: str) -> None:
        if sid != self.webrtc.sid or profile != self.profile:
            return
        await self.webrtc.end_call()
        if self.on_call_ended_cb is not None:
            await aio.maybe_async(self.on_call_ended_cb(sid, profile))

    async def start(self):
        """Start a call.

        To be used only if we are initiator
        """
        try:
            await self.webrtc.setup_call("initiator")
        except Exception:
            log.exception("Can't setup call")
            raise
        self.webrtc.start_pipeline()