diff options
Diffstat (limited to 'libdino/src/service/calls.vala')
| -rw-r--r-- | libdino/src/service/calls.vala | 686 |
1 files changed, 686 insertions, 0 deletions
diff --git a/libdino/src/service/calls.vala b/libdino/src/service/calls.vala new file mode 100644 index 00000000..4c3bbea7 --- /dev/null +++ b/libdino/src/service/calls.vala @@ -0,0 +1,686 @@ +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 encryption_updated(Call call, Xep.Jingle.ContentEncryption? audio_encryption, Xep.Jingle.ContentEncryption? video_encryption, bool same); + + 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 Database db; + + 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 HashMap<Account, Call> jmi_call = new HashMap<Account, Call>(Account.hash_func, Account.equals_func); + public HashMap<Account, string> jmi_sid = new HashMap<Account, string>(Account.hash_func, Account.equals_func); + public HashMap<Account, bool> jmi_video = new HashMap<Account, bool>(Account.hash_func, Account.equals_func); + + 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> audio_content = new HashMap<Call, Xep.Jingle.Content>(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); + private HashMap<Call, HashMap<string, Xep.Jingle.ContentEncryption>> video_encryptions = new HashMap<Call, HashMap<string, Xep.Jingle.ContentEncryption>>(Call.hash_func, Call.equals_func); + private HashMap<Call, HashMap<string, Xep.Jingle.ContentEncryption>> audio_encryptions = new HashMap<Call, HashMap<string, Xep.Jingle.ContentEncryption>>(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; + this.db = db; + + 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 = call.end_time = new DateTime.now_utc(); + call.state = Call.State.RINGING; + + stream_interactor.get_module(CallStore.IDENTITY).add_call(call, conversation); + + we_should_send_video[call] = video; + we_should_send_audio[call] = true; + + Gee.List<Jid> call_resources = yield get_call_resources(conversation); + + bool do_jmi = false; + Jid? jid_for_direct = null; + if (yield contains_jmi_resources(conversation.account, call_resources)) { + do_jmi = true; + } else if (!call_resources.is_empty) { + jid_for_direct = call_resources[0]; + } else if (has_jmi_resources(conversation)) { + do_jmi = true; + } + + if (do_jmi) { + XmppStream? stream = stream_interactor.get_stream(conversation.account); + jmi_call[conversation.account] = call; + jmi_video[conversation.account] = video; + jmi_sid[conversation.account] = Xmpp.random_uuid(); + + call_by_sid[call.account][jmi_sid[conversation.account]] = call; + + var descriptions = new ArrayList<StanzaNode>(); + descriptions.add(new StanzaNode.build("description", Xep.JingleRtp.NS_URI).add_self_xmlns().put_attribute("media", "audio")); + if (video) { + descriptions.add(new StanzaNode.build("description", Xep.JingleRtp.NS_URI).add_self_xmlns().put_attribute("media", "video")); + } + + stream.get_module(Xmpp.Xep.JingleMessageInitiation.Module.IDENTITY).send_session_propose_to_peer(stream, conversation.counterpart, jmi_sid[call.account], descriptions); + } else if (jid_for_direct != null) { + yield call_resource(conversation.account, jid_for_direct, call, video); + } + + conversation.last_active = call.time; + call_outgoing(call, conversation); + + return call; + } + + private async void call_resource(Account account, Jid full_jid, Call call, bool video, string? sid = null) { + XmppStream? stream = stream_interactor.get_stream(account); + if (stream == null) return; + + Xep.Jingle.Session session = yield stream.get_module(Xep.JingleRtp.Module.IDENTITY).start_call(stream, full_jid, video, sid); + sessions[call] = session; + sid_by_call[call.account][call] = session.sid; + + connect_session_signals(call, session); + } + + 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 + stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_retract_to_peer(stream, call.counterpart, jmi_sid[call.account]); + } + 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) { + content.accept(); + } + } else { + // Only a JMI so far + Account account = call.account; + string sid = sid_by_call[call.account][call]; + XmppStream stream = stream_interactor.get_stream(account); + if (stream == null) return; + + jmi_call[account] = call; + jmi_sid[account] = sid; + jmi_video[account] = we_should_send_video[call]; + + stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_accept_to_self(stream, sid); + stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_proceed_to_peer(stream, call.counterpart, 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) { + 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_peer(stream, call.counterpart, sid); + stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_reject_to_self(stream, 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; + + if (!sessions.has_key(call)) { + // Call hasn't been established yet + return; + } + + 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 bool can_do_audio_calls_async(Conversation conversation) { + if (!can_do_audio_calls()) return false; + return (yield get_call_resources(conversation)).size > 0 || has_jmi_resources(conversation); + } + + private bool can_do_audio_calls() { + Plugins.VideoCallPlugin? plugin = Application.get_default().plugin_registry.video_call_plugin; + if (plugin == null) return false; + + return plugin.supports("audio"); + } + + public async bool can_do_video_calls_async(Conversation conversation) { + if (!can_do_video_calls()) return false; + return (yield get_call_resources(conversation)).size > 0 || has_jmi_resources(conversation); + } + + private bool can_do_video_calls() { + Plugins.VideoCallPlugin? plugin = Application.get_default().plugin_registry.video_call_plugin; + if (plugin == null) return false; + + return plugin.supports("video"); + } + + private 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; + ret.add(full_jid); + } + return ret; + } + + private async bool contains_jmi_resources(Account account, Gee.List<Jid> full_jids) { + XmppStream? stream = stream_interactor.get_stream(account); + if (stream == null) return false; + + foreach (Jid full_jid in full_jids) { + bool does_jmi = yield stream_interactor.get_module(EntityInfo.IDENTITY).has_feature(account, full_jid, Xep.JingleMessageInitiation.NS_URI); + if (does_jmi) return true; + } + return false; + } + + private bool has_jmi_resources(Conversation conversation) { + int64 jmi_resources = db.entity.select() + .with(db.entity.jid_id, "=", db.get_jid_id(conversation.counterpart)) + .join_with(db.entity_feature, db.entity.caps_hash, db.entity_feature.entity) + .with(db.entity_feature.feature, "=", Xep.JingleMessageInitiation.NS_URI) + .count(); + return jmi_resources > 0; + } + + 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) { + if (!can_do_audio_calls()) { + warning("Incoming call but no call support detected. Ignoring."); + return; + } + + bool counterpart_wants_video = false; + foreach (Xep.Jingle.Content content in session.contents) { + 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 = jmi_sid.has_key(account) && + jmi_sid[account] == session.sid && jmi_call[account].account.equals(account) && + jmi_call[account].counterpart.equals_bare(session.peer_full_jid) && + jmi_video[account] == counterpart_wants_video; + + Call? call = null; + if (already_accepted) { + call = jmi_call[account]; + } 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 = call.end_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) { + content.reject(); + return; + } + + // Our peer shouldn't tell us to start sending, that's for us to initiate + if (session.senders_include_us(content.senders)) { + if (session.senders_include_counterpart(content.senders)) { + // If our peer wants to send, let them + content.modify(session.we_initiated ? Xep.Jingle.Senders.RESPONDER : Xep.Jingle.Senders.INITIATOR); + } else { + // If only we're supposed to send, reject + content.reject(); + } + } + + connect_content_signals(call, content, rtp_content_parameter); + content.accept(); + } + + 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; + } 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) { + 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[call] = content; + 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, rtp_content_parameter.media)); + + 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 on_connection_ready(Call call, Xep.Jingle.Content content, string media) { + if (call.state == Call.State.RINGING || call.state == Call.State.ESTABLISHING) { + call.state = Call.State.IN_PROGRESS; + } + + if (media == "audio") { + audio_encryptions[call] = content.encryptions; + } else if (media == "video") { + video_encryptions[call] = content.encryptions; + } + + if ((audio_encryptions.has_key(call) && audio_encryptions[call].is_empty) || (video_encryptions.has_key(call) && video_encryptions[call].is_empty)) { + call.encryption = Encryption.NONE; + encryption_updated(call, null, null, true); + return; + } + + HashMap<string, Xep.Jingle.ContentEncryption> encryptions = audio_encryptions[call] ?? video_encryptions[call]; + + Xep.Jingle.ContentEncryption? omemo_encryption = null, dtls_encryption = null, srtp_encryption = null; + foreach (string encr_name in encryptions.keys) { + if (video_encryptions.has_key(call) && !video_encryptions[call].has_key(encr_name)) continue; + + var encryption = encryptions[encr_name]; + if (encryption.encryption_ns == "http://gultsch.de/xmpp/drafts/omemo/dlts-srtp-verification") { + omemo_encryption = encryption; + } else if (encryption.encryption_ns == Xep.JingleIceUdp.DTLS_NS_URI) { + dtls_encryption = encryption; + } else if (encryption.encryption_name == "SRTP") { + srtp_encryption = encryption; + } + } + + if (omemo_encryption != null && dtls_encryption != null) { + call.encryption = Encryption.OMEMO; + Xep.Jingle.ContentEncryption? video_encryption = video_encryptions.has_key(call) ? video_encryptions[call]["http://gultsch.de/xmpp/drafts/omemo/dlts-srtp-verification"] : null; + omemo_encryption.peer_key = dtls_encryption.peer_key; + omemo_encryption.our_key = dtls_encryption.our_key; + encryption_updated(call, omemo_encryption, video_encryption, true); + } else if (dtls_encryption != null) { + call.encryption = Encryption.DTLS_SRTP; + Xep.Jingle.ContentEncryption? video_encryption = video_encryptions.has_key(call) ? video_encryptions[call][Xep.JingleIceUdp.DTLS_NS_URI] : null; + bool same = true; + if (video_encryption != null && dtls_encryption.peer_key.length == video_encryption.peer_key.length) { + for (int i = 0; i < dtls_encryption.peer_key.length; i++) { + if (dtls_encryption.peer_key[i] != video_encryption.peer_key[i]) { same = false; break; } + } + } + encryption_updated(call, dtls_encryption, video_encryption, same); + } else if (srtp_encryption != null) { + call.encryption = Encryption.SRTP; + encryption_updated(call, srtp_encryption, video_encryptions[call]["SRTP"], false); + } else { + call.encryption = Encryption.NONE; + encryption_updated(call, null, null, true); + } + } + + 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); + audio_content.unset(call); + video_content.unset(call); + audio_encryptions.unset(call); + video_encryptions.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) { + 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) { + 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) => { + if (!can_do_audio_calls()) { + warning("Incoming call but no call support detected. Ignoring."); + return; + } + + 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; + + if (from.equals_bare(account.bare_jid)) { // Carboned message from our account + // Ignore carbon from ourselves + if (from.equals(account.full_jid)) return; + + Call call = call_by_sid[account][sid]; + call.state = Call.State.OTHER_DEVICE_ACCEPTED; + remove_call_from_datastructures(call); + } else if (from.equals_bare(call_by_sid[account][sid].counterpart)) { // Message from our peer + // We proposed the call + if (jmi_sid.has_key(account) && jmi_sid[account] == sid) { + call_resource.begin(account, from, jmi_call[account], jmi_video[account], jmi_sid[account]); + jmi_call.unset(account); + jmi_sid.unset(account); + jmi_video.unset(account); + } + } + }); + mi_module.session_rejected.connect((from, to, sid) => { + if (!call_by_sid[account].has_key(sid)) return; + Call call = call_by_sid[account][sid]; + + bool outgoing_reject = call.direction == Call.DIRECTION_OUTGOING && from.equals_bare(call.counterpart); + bool incoming_reject = call.direction == Call.DIRECTION_INCOMING && from.equals_bare(account.bare_jid); + if (!(outgoing_reject || incoming_reject)) return; + + 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]; + + bool outgoing_retract = call.direction == Call.DIRECTION_OUTGOING && from.equals_bare(call.counterpart); + bool incoming_retract = call.direction == Call.DIRECTION_INCOMING && from.equals_bare(account.bare_jid); + if (!(outgoing_retract || incoming_retract)) return; + + call.state = Call.State.MISSED; + remove_call_from_datastructures(call); + call_terminated(call, null, null); + }); + } + } +}
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