Mercurial > libervia-backend
view libervia/cli/call_webrtc.py @ 4231:e11b13418ba6
plugin XEP-0353, XEP-0234, jingle: WebRTC data channel signaling implementation:
Implement XEP-0343: Signaling WebRTC Data Channels in Jingle. The current version of the
XEP (0.3.1) has no implementation and contains some flaws. After discussing this on xsf@,
Daniel (from Conversations) mentioned that they had a sprint with Larma (from Dino) to
work on another version and provided me with this link:
https://gist.github.com/iNPUTmice/6c56f3e948cca517c5fb129016d99e74 . I have used it for my
implementation.
This implementation reuses work done on Jingle A/V call (notably XEP-0176 and XEP-0167
plugins), with adaptations. When used, XEP-0234 will not handle the file itself as it
normally does. This is because WebRTC has several implementations (browser for web
interface, GStreamer for others), and file/data must be handled directly by the frontend.
This is particularly important for web frontends, as the file is not sent from the backend
but from the end-user's browser device.
Among the changes, there are:
- XEP-0343 implementation.
- `file_send` bridge method now use serialised dict as output.
- New `BaseTransportHandler.is_usable` method which get content data and returns a boolean
(default to `True`) to tell if this transport can actually be used in this context (when
we are initiator). Used in webRTC case to see if call data are available.
- Support of `application` media type, and everything necessary to handle data channels.
- Better confirmation message, with file name, size and description when available.
- When file is accepted in preflight, it is specified in following `action_new` signal for
actual file transfer. This way, frontend can avoid the display or 2 confirmation
messages.
- XEP-0166: when not specified, default `content` name is now its index number instead of
a UUID. This follows the behaviour of browsers.
- XEP-0353: better handling of events such as call taken by another device.
- various other updates.
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author | Goffi <goffi@goffi.org> |
---|---|
date | Sat, 06 Apr 2024 12:57:23 +0200 |
parents | 9218d4331bb2 |
children |
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#!/usr/bin/env python3 # Libervia CLI # Copyright (C) 2009-2024 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 dataclasses import dataclass from libervia.backend.tools.common import data_format from libervia.frontends.tools import aio, jid @dataclass class CallData: callee: jid.JID sid: str | None = None action_id: str | None = None class WebRTCCall: def __init__(self, host, profile: str, callee: jid.JID, **kwargs): """Create and setup a webRTC instance @param profile: profile making or receiving the call @param callee: peer jid @param kwargs: extra kw args to use when instantiating WebRTC """ from libervia.frontends.tools import webrtc aio.install_glib_asyncio_iteration() self.host = host self.profile = profile self.webrtc = webrtc.WebRTC(host.bridge, profile, **kwargs) self.webrtc.callee = callee host.bridge.register_signal( "ice_candidates_new", self.on_ice_candidates_new, "plugin" ) host.bridge.register_signal("call_setup", self.on_call_setup, "plugin") host.bridge.register_signal("call_ended", self.on_call_ended, "plugin") @classmethod async def make_webrtc_call( cls, host, 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(host, profile, call_data.callee, **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 host.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: self.host.disp(f"Invalid setup data received: {setup_data}", error=True) return if role == "initiator": self.webrtc.on_accepted_call(sdp, profile) elif role == "responder": await self.webrtc.answer_call(sdp, profile) else: self.host.disp( f"Invalid role received during setup: {setup_data}", error=True ) # we want to be sure that call is ended if user presses `Ctrl + c` or anything # else stops the session. self.host.add_on_quit_callback( lambda: self.host.bridge.call_end(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() await self.host.a_quit() async def start(self): """Start a call. To be used only if we are initiator """ await self.webrtc.setup_call("initiator") self.webrtc.start_pipeline()