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authorfiaxh <git@lightrise.org>2021-03-19 23:07:40 +0100
committerfiaxh <git@lightrise.org>2021-03-21 12:41:39 +0100
commitcdb4d77259e6c361aaca64a483a43d7441f4803d (patch)
tree6db053f9f0875f54261f3f7e7d6830612076e29d /libdino/src/service/calls.vala
parentef2e3c774cab82a94a5e34399f2013d64c3cf03b (diff)
downloaddino-cdb4d77259e6c361aaca64a483a43d7441f4803d.tar.gz
dino-cdb4d77259e6c361aaca64a483a43d7441f4803d.zip
Add support for unencrypted RTP calls to libdino
Co-authored-by: Marvin W <git@larma.de>
Diffstat (limited to 'libdino/src/service/calls.vala')
-rw-r--r--libdino/src/service/calls.vala514
1 files changed, 514 insertions, 0 deletions
diff --git a/libdino/src/service/calls.vala b/libdino/src/service/calls.vala
new file mode 100644
index 00000000..5224bdd1
--- /dev/null
+++ b/libdino/src/service/calls.vala
@@ -0,0 +1,514 @@
+using Gee;
+
+using Xmpp;
+using Dino.Entities;
+
+namespace Dino {
+
+ public class Calls : StreamInteractionModule, Object {
+
+ public signal void call_incoming(Call call, Conversation conversation, bool video);
+ public signal void call_outgoing(Call call, Conversation conversation);
+
+ public signal void call_terminated(Call call, string? reason_name, string? reason_text);
+ public signal void counterpart_ringing(Call call);
+ public signal void counterpart_sends_video_updated(Call call, bool mute);
+ public signal void info_received(Call call, Xep.JingleRtp.CallSessionInfo session_info);
+
+ public signal void stream_created(Call call, string media);
+
+ public static ModuleIdentity<Calls> IDENTITY = new ModuleIdentity<Calls>("calls");
+ public string id { get { return IDENTITY.id; } }
+
+ private StreamInteractor stream_interactor;
+ private Xep.JingleRtp.SessionInfoType session_info_type;
+
+ private HashMap<Account, HashMap<Call, string>> sid_by_call = new HashMap<Account, HashMap<Call, string>>(Account.hash_func, Account.equals_func);
+ private HashMap<Account, HashMap<string, Call>> call_by_sid = new HashMap<Account, HashMap<string, Call>>(Account.hash_func, Account.equals_func);
+ public HashMap<Call, Xep.Jingle.Session> sessions = new HashMap<Call, Xep.Jingle.Session>(Call.hash_func, Call.equals_func);
+
+ public Call? mi_accepted_call = null;
+ public string? mi_accepted_sid = null;
+ public bool mi_accepted_video = false;
+
+ private HashMap<Call, bool> counterpart_sends_video = new HashMap<Call, bool>(Call.hash_func, Call.equals_func);
+ private HashMap<Call, bool> we_should_send_video = new HashMap<Call, bool>(Call.hash_func, Call.equals_func);
+ private HashMap<Call, bool> we_should_send_audio = new HashMap<Call, bool>(Call.hash_func, Call.equals_func);
+
+ private HashMap<Call, Xep.JingleRtp.Parameters> audio_content_parameter = new HashMap<Call, Xep.JingleRtp.Parameters>(Call.hash_func, Call.equals_func);
+ private HashMap<Call, Xep.JingleRtp.Parameters> video_content_parameter = new HashMap<Call, Xep.JingleRtp.Parameters>(Call.hash_func, Call.equals_func);
+ private HashMap<Call, Xep.Jingle.Content> video_content = new HashMap<Call, Xep.Jingle.Content>(Call.hash_func, Call.equals_func);
+
+ public static void start(StreamInteractor stream_interactor, Database db) {
+ Calls m = new Calls(stream_interactor, db);
+ stream_interactor.add_module(m);
+ }
+
+ private Calls(StreamInteractor stream_interactor, Database db) {
+ this.stream_interactor = stream_interactor;
+
+ stream_interactor.account_added.connect(on_account_added);
+ }
+
+ public Xep.JingleRtp.Stream? get_video_stream(Call call) {
+ if (video_content_parameter.has_key(call)) {
+ return video_content_parameter[call].stream;
+ }
+ return null;
+ }
+
+ public Xep.JingleRtp.Stream? get_audio_stream(Call call) {
+ if (audio_content_parameter.has_key(call)) {
+ return audio_content_parameter[call].stream;
+ }
+ return null;
+ }
+
+ public async Call? initiate_call(Conversation conversation, bool video) {
+ Call call = new Call();
+ call.direction = Call.DIRECTION_OUTGOING;
+ call.account = conversation.account;
+ call.counterpart = conversation.counterpart;
+ call.ourpart = conversation.account.full_jid;
+ call.time = call.local_time = new DateTime.now_utc();
+ call.state = Call.State.RINGING;
+
+ stream_interactor.get_module(CallStore.IDENTITY).add_call(call, conversation);
+
+ XmppStream? stream = stream_interactor.get_stream(conversation.account);
+ if (stream == null) return null;
+
+ Gee.List<Jid> call_resources = yield get_call_resources(conversation);
+ if (call_resources.size > 0) {
+ Jid full_jid = call_resources[0];
+ Xep.Jingle.Session session = yield stream.get_module(Xep.JingleRtp.Module.IDENTITY).start_call(stream, full_jid, video);
+ sessions[call] = session;
+ call_by_sid[call.account][session.sid] = call;
+ sid_by_call[call.account][call] = session.sid;
+
+ connect_session_signals(call, session);
+ }
+
+ we_should_send_video[call] = video;
+ we_should_send_audio[call] = true;
+
+ conversation.last_active = call.time;
+ call_outgoing(call, conversation);
+
+ return call;
+ }
+
+ public void end_call(Conversation conversation, Call call) {
+ XmppStream? stream = stream_interactor.get_stream(call.account);
+ if (stream == null) return;
+
+ if (call.state == Call.State.IN_PROGRESS || call.state == Call.State.ESTABLISHING) {
+ sessions[call].terminate(Xep.Jingle.ReasonElement.SUCCESS, null, "success");
+ call.state = Call.State.ENDED;
+ } else if (call.state == Call.State.RINGING) {
+ if (sessions.has_key(call)) {
+ sessions[call].terminate(Xep.Jingle.ReasonElement.CANCEL, null, "cancel");
+ } else {
+ // Only a JMI so far
+ }
+ call.state = Call.State.MISSED;
+ } else {
+ return;
+ }
+
+ call.end_time = new DateTime.now_utc();
+
+ remove_call_from_datastructures(call);
+ }
+
+ public void accept_call(Call call) {
+ call.state = Call.State.ESTABLISHING;
+
+ if (sessions.has_key(call)) {
+ foreach (Xep.Jingle.Content content in sessions[call].contents.values) {
+ content.accept();
+ }
+ } else {
+ // Only a JMI so far
+ XmppStream stream = stream_interactor.get_stream(call.account);
+ if (stream == null) return;
+
+ mi_accepted_call = call;
+ mi_accepted_sid = sid_by_call[call.account][call];
+ mi_accepted_video = we_should_send_video[call];
+
+ stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_accept_to_self(stream, mi_accepted_sid);
+ stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_proceed_to_peer(stream, call.counterpart, mi_accepted_sid);
+ }
+ }
+
+ public void reject_call(Call call) {
+ call.state = Call.State.DECLINED;
+
+ if (sessions.has_key(call)) {
+ foreach (Xep.Jingle.Content content in sessions[call].contents.values) {
+ content.reject();
+ }
+ remove_call_from_datastructures(call);
+ } else {
+ // Only a JMI so far
+ XmppStream stream = stream_interactor.get_stream(call.account);
+ if (stream == null) return;
+
+ string sid = sid_by_call[call.account][call];
+ stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_reject_to_self(stream, sid);
+ stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_reject_to_peer(stream, call.counterpart, sid);
+ remove_call_from_datastructures(call);
+ }
+ }
+
+ public void mute_own_audio(Call call, bool mute) {
+ we_should_send_audio[call] = !mute;
+
+ Xep.JingleRtp.Stream stream = audio_content_parameter[call].stream;
+ // The user might mute audio before a feed was created. The feed will be muted as soon as it has been created.
+ if (stream == null) return;
+
+ // Inform our counterpart that we (un)muted our audio
+ stream_interactor.module_manager.get_module(call.account, Xep.JingleRtp.Module.IDENTITY).session_info_type.send_mute(sessions[call], mute, "audio");
+
+ // Start/Stop sending audio data
+ Application.get_default().plugin_registry.video_call_plugin.set_pause(stream, mute);
+ }
+
+ public void mute_own_video(Call call, bool mute) {
+ we_should_send_video[call] = !mute;
+
+ Xep.JingleRtp.Module rtp_module = stream_interactor.module_manager.get_module(call.account, Xep.JingleRtp.Module.IDENTITY);
+
+ if (video_content_parameter.has_key(call) &&
+ video_content_parameter[call].stream != null &&
+ sessions[call].senders_include_us(video_content[call].senders)) {
+ // A video feed has already been established
+
+ // Start/Stop sending video data
+ Xep.JingleRtp.Stream stream = video_content_parameter[call].stream;
+ if (stream != null) {
+ // TODO maybe the user muted video before the feed was created...
+ Application.get_default().plugin_registry.video_call_plugin.set_pause(stream, mute);
+ }
+
+ // Inform our counterpart that we started/stopped our video
+ rtp_module.session_info_type.send_mute(sessions[call], mute, "video");
+ } else if (!mute) {
+ // Need to start a new video feed
+ XmppStream stream = stream_interactor.get_stream(call.account);
+ rtp_module.add_outgoing_video_content.begin(stream, sessions[call], (_, res) => {
+ if (video_content_parameter[call] == null) {
+ Xep.Jingle.Content content = rtp_module.add_outgoing_video_content.end(res);
+ Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
+ if (rtp_content_parameter != null) {
+ connect_content_signals(call, content, rtp_content_parameter);
+ }
+ }
+ });
+ }
+ // If video_feed == null && !mute we're trying to mute a non-existant feed. It will be muted as soon as it is created.
+ }
+
+ public async Gee.List<Jid> get_call_resources(Conversation conversation) {
+ ArrayList<Jid> ret = new ArrayList<Jid>();
+
+ XmppStream? stream = stream_interactor.get_stream(conversation.account);
+ if (stream == null) return ret;
+
+ Gee.List<Jid>? full_jids = stream.get_flag(Presence.Flag.IDENTITY).get_resources(conversation.counterpart);
+ if (full_jids == null) return ret;
+
+ foreach (Jid full_jid in full_jids) {
+ bool supports_rtc = yield stream.get_module(Xep.JingleRtp.Module.IDENTITY).is_available(stream, full_jid);
+ if (!supports_rtc) continue;
+
+ // dtls support indicates webRTC support. Clients tend to not do normal ice udp in that case. Except Dino.
+ bool supports_dtls = yield stream_interactor.get_module(EntityInfo.IDENTITY).has_feature(conversation.account, full_jid, "urn:xmpp:jingle:apps:dtls:0");
+ if (supports_dtls) {
+ Xep.ServiceDiscovery.Identity? identity = yield stream_interactor.get_module(EntityInfo.IDENTITY).get_identity(conversation.account, full_jid);
+ bool is_dino = identity != null && identity.name == "Dino";
+
+ if (!is_dino) continue;
+ }
+
+ ret.add(full_jid);
+ }
+ return ret;
+ }
+
+ public bool should_we_send_video(Call call) {
+ return we_should_send_video[call];
+ }
+
+ public Jid? is_call_in_progress() {
+ foreach (Call call in sessions.keys) {
+ if (call.state == Call.State.IN_PROGRESS || call.state == Call.State.RINGING || call.state == Call.State.ESTABLISHING) {
+ return call.counterpart;
+ }
+ }
+ return null;
+ }
+
+ private void on_incoming_call(Account account, Xep.Jingle.Session session) {
+ bool counterpart_wants_video = false;
+ foreach (Xep.Jingle.Content content in session.contents.values) {
+ Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
+ if (rtp_content_parameter == null) continue;
+ if (rtp_content_parameter.media == "video" && session.senders_include_us(content.senders)) {
+ counterpart_wants_video = true;
+ }
+ }
+
+ // Session might have already been accepted via Jingle Message Initiation
+ bool already_accepted = mi_accepted_sid == session.sid && mi_accepted_call.account.equals(account) &&
+ mi_accepted_call.counterpart.equals_bare(session.peer_full_jid) &&
+ mi_accepted_video == counterpart_wants_video;
+
+ Call? call = null;
+ if (already_accepted) {
+ call = mi_accepted_call;
+ } else {
+ call = create_received_call(account, session.peer_full_jid, account.full_jid, counterpart_wants_video);
+ }
+ sessions[call] = session;
+
+ call_by_sid[account][session.sid] = call;
+ sid_by_call[account][call] = session.sid;
+
+ connect_session_signals(call, session);
+
+ if (already_accepted) {
+ accept_call(call);
+ } else {
+ stream_interactor.module_manager.get_module(account, Xep.JingleRtp.Module.IDENTITY).session_info_type.send_ringing(session);
+ }
+ }
+
+ private Call create_received_call(Account account, Jid from, Jid to, bool video_requested) {
+ Call call = new Call();
+ if (from.equals_bare(account.bare_jid)) {
+ // Call requested by another of our devices
+ call.direction = Call.DIRECTION_OUTGOING;
+ call.ourpart = from;
+ call.counterpart = to;
+ } else {
+ call.direction = Call.DIRECTION_INCOMING;
+ call.ourpart = account.full_jid;
+ call.counterpart = from;
+ }
+ call.account = account;
+ call.time = call.local_time = new DateTime.now_utc();
+ call.state = Call.State.RINGING;
+
+ Conversation conversation = stream_interactor.get_module(ConversationManager.IDENTITY).create_conversation(call.counterpart.bare_jid, account, Conversation.Type.CHAT);
+
+ stream_interactor.get_module(CallStore.IDENTITY).add_call(call, conversation);
+
+ conversation.last_active = call.time;
+
+ we_should_send_video[call] = video_requested;
+ we_should_send_audio[call] = true;
+
+ if (call.direction == Call.DIRECTION_INCOMING) {
+ call_incoming(call, conversation, video_requested);
+ } else {
+ call_outgoing(call, conversation);
+ }
+
+ return call;
+ }
+
+ private void on_incoming_content_add(XmppStream stream, Call call, Xep.Jingle.Session session, Xep.Jingle.Content content) {
+ Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
+
+ if (rtp_content_parameter == null || session.senders_include_us(content.senders)) {
+ content.reject();
+ return;
+ }
+
+ connect_content_signals(call, content, rtp_content_parameter);
+ content.accept();
+ }
+
+ private void on_connection_ready(Call call) {
+ if (call.state == Call.State.RINGING || call.state == Call.State.ESTABLISHING) {
+ call.state = Call.State.IN_PROGRESS;
+ }
+ }
+
+ private void on_call_terminated(Call call, bool we_terminated, string? reason_name, string? reason_text) {
+ if (call.state == Call.State.RINGING || call.state == Call.State.IN_PROGRESS || call.state == Call.State.ESTABLISHING) {
+ call.end_time = new DateTime.now_utc();
+ }
+ if (call.state == Call.State.IN_PROGRESS) {
+ call.state = Call.State.ENDED;
+ call_terminated(call, reason_name, reason_text);
+ } else if (call.state == Call.State.RINGING || call.state == Call.State.ESTABLISHING) {
+ if (reason_name == Xep.Jingle.ReasonElement.DECLINE) {
+ call.state = Call.State.DECLINED;
+ } else {
+ call.state = Call.State.FAILED;
+ }
+ call_terminated(call, reason_name, reason_text);
+ }
+
+ remove_call_from_datastructures(call);
+ }
+
+ private void on_stream_created(Call call, string media, Xep.JingleRtp.Stream stream) {
+ if (media == "video" && stream.receiving) {
+ counterpart_sends_video[call] = true;
+ video_content_parameter[call].connection_ready.connect((status) => {
+ counterpart_sends_video_updated(call, false);
+ });
+ }
+ stream_created(call, media);
+
+ // Outgoing audio/video might have been muted in the meanwhile.
+ if (media == "video" && !we_should_send_video[call]) {
+ mute_own_video(call, true);
+ } else if (media == "audio" && !we_should_send_audio[call]) {
+ mute_own_audio(call, true);
+ }
+ }
+
+ private void on_counterpart_mute_update(Call call, bool mute, string? media) {
+ if (!call.equals(call)) return;
+
+ if (media == "video") {
+ counterpart_sends_video[call] = !mute;
+ counterpart_sends_video_updated(call, mute);
+ }
+ }
+
+ private void connect_session_signals(Call call, Xep.Jingle.Session session) {
+ session.terminated.connect((stream, we_terminated, reason_name, reason_text) =>
+ on_call_terminated(call, we_terminated, reason_name, reason_text)
+ );
+ session.additional_content_add_incoming.connect((session,stream, content) =>
+ on_incoming_content_add(stream, call, session, content)
+ );
+
+ foreach (Xep.Jingle.Content content in session.contents.values) {
+ Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
+ if (rtp_content_parameter == null) continue;
+
+ connect_content_signals(call, content, rtp_content_parameter);
+ }
+ }
+
+ private void connect_content_signals(Call call, Xep.Jingle.Content content, Xep.JingleRtp.Parameters rtp_content_parameter) {
+ if (rtp_content_parameter.media == "audio") {
+ audio_content_parameter[call] = rtp_content_parameter;
+ } else if (rtp_content_parameter.media == "video") {
+ video_content[call] = content;
+ video_content_parameter[call] = rtp_content_parameter;
+ }
+
+ rtp_content_parameter.stream_created.connect((stream) => on_stream_created(call, rtp_content_parameter.media, stream));
+ rtp_content_parameter.connection_ready.connect((status) => on_connection_ready(call));
+
+ content.senders_modify_incoming.connect((content, proposed_senders) => {
+ if (content.session.senders_include_us(content.senders) != content.session.senders_include_us(proposed_senders)) {
+ warning("counterpart set us to (not)sending %s. ignoring", content.content_name);
+ return;
+ }
+
+ if (!content.session.senders_include_counterpart(content.senders) && content.session.senders_include_counterpart(proposed_senders)) {
+ // Counterpart wants to start sending. Ok.
+ content.accept_content_modify(proposed_senders);
+ on_counterpart_mute_update(call, false, "video");
+ }
+ });
+ }
+
+ private void remove_call_from_datastructures(Call call) {
+ string? sid = sid_by_call[call.account][call];
+ sid_by_call[call.account].unset(call);
+ if (sid != null) call_by_sid[call.account].unset(sid);
+
+ sessions.unset(call);
+
+ counterpart_sends_video.unset(call);
+ we_should_send_video.unset(call);
+ we_should_send_audio.unset(call);
+
+ audio_content_parameter.unset(call);
+ video_content_parameter.unset(call);
+ video_content.unset(call);
+ }
+
+ private void on_account_added(Account account) {
+ call_by_sid[account] = new HashMap<string, Call>();
+ sid_by_call[account] = new HashMap<Call, string>();
+
+ Xep.Jingle.Module jingle_module = stream_interactor.module_manager.get_module(account, Xep.Jingle.Module.IDENTITY);
+ jingle_module.session_initiate_received.connect((stream, session) => {
+ foreach (Xep.Jingle.Content content in session.contents.values) {
+ Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
+ if (rtp_content_parameter != null) {
+ on_incoming_call(account, session);
+ break;
+ }
+ }
+ });
+
+ var session_info_type = stream_interactor.module_manager.get_module(account, Xep.JingleRtp.Module.IDENTITY).session_info_type;
+ session_info_type.mute_update_received.connect((session,mute, name) => {
+ if (!call_by_sid[account].has_key(session.sid)) return;
+ Call call = call_by_sid[account][session.sid];
+
+ foreach (Xep.Jingle.Content content in session.contents.values) {
+ if (name == null || content.content_name == name) {
+ Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
+ if (rtp_content_parameter != null) {
+ on_counterpart_mute_update(call, mute, rtp_content_parameter.media);
+ }
+ }
+ }
+ });
+ session_info_type.info_received.connect((session, session_info) => {
+ if (!call_by_sid[account].has_key(session.sid)) return;
+ Call call = call_by_sid[account][session.sid];
+
+ info_received(call, session_info);
+ });
+
+ Xep.JingleMessageInitiation.Module mi_module = stream_interactor.module_manager.get_module(account, Xep.JingleMessageInitiation.Module.IDENTITY);
+ mi_module.session_proposed.connect((from, to, sid, descriptions) => {
+ bool audio_requested = descriptions.any_match((description) => description.ns_uri == Xep.JingleRtp.NS_URI && description.get_attribute("media") == "audio");
+ bool video_requested = descriptions.any_match((description) => description.ns_uri == Xep.JingleRtp.NS_URI && description.get_attribute("media") == "video");
+ if (!audio_requested && !video_requested) return;
+ Call call = create_received_call(account, from, to, video_requested);
+ call_by_sid[account][sid] = call;
+ sid_by_call[account][call] = sid;
+ });
+ mi_module.session_accepted.connect((from, sid) => {
+ if (!call_by_sid[account].has_key(sid)) return;
+
+ // Ignore session-accepted from ourselves
+ if (!from.equals(account.full_jid)) {
+ Call call = call_by_sid[account][sid];
+ call.state = Call.State.OTHER_DEVICE_ACCEPTED;
+ remove_call_from_datastructures(call);
+ }
+ });
+ mi_module.session_rejected.connect((from, to, sid) => {
+ if (!call_by_sid[account].has_key(sid)) return;
+ Call call = call_by_sid[account][sid];
+ call.state = Call.State.DECLINED;
+ remove_call_from_datastructures(call);
+ call_terminated(call, null, null);
+ });
+ mi_module.session_retracted.connect((from, to, sid) => {
+ if (!call_by_sid[account].has_key(sid)) return;
+ Call call = call_by_sid[account][sid];
+ call.state = Call.State.MISSED;
+ remove_call_from_datastructures(call);
+ call_terminated(call, null, null);
+ });
+ }
+ }
+} \ No newline at end of file