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using Dino.Entities;
using Gee;
using Xmpp;
public class Dino.PeerState : Object {
public signal void stream_created(string media);
public signal void counterpart_sends_video_updated(bool mute);
public signal void info_received(Xep.JingleRtp.CallSessionInfo session_info);
public signal void connection_ready();
public signal void session_terminated(bool we_terminated, string? reason_name, string? reason_text);
public signal void encryption_updated(Xep.Jingle.ContentEncryption? audio_encryption, Xep.Jingle.ContentEncryption? video_encryption, bool same);
public StreamInteractor stream_interactor;
CallState call_state;
public Calls calls;
public Call call;
public Jid jid;
public Xep.Jingle.Session session;
public string sid;
public string internal_id = Xmpp.random_uuid();
public Xep.JingleRtp.Parameters? audio_content_parameter = null;
public Xep.JingleRtp.Parameters? video_content_parameter = null;
public Xep.Jingle.Content? audio_content = null;
public Xep.Jingle.Content? video_content = null;
public Xep.Jingle.ContentEncryption? video_encryption = null;
public Xep.Jingle.ContentEncryption? audio_encryption = null;
public bool encryption_keys_same = false;
public HashMap<string, Xep.Jingle.ContentEncryption>? video_encryptions = null;
public HashMap<string, Xep.Jingle.ContentEncryption>? audio_encryptions = null;
public bool first_peer = false;
public bool waiting_for_inbound_muji_connection = false;
public Xep.Muji.GroupCall? group_call { get; set; }
public bool counterpart_sends_video = false;
public bool we_should_send_audio { get; set; default=false; }
public bool we_should_send_video { get; set; default=false; }
public PeerState(Jid jid, Call call, CallState call_state, StreamInteractor stream_interactor) {
this.jid = jid;
this.call = call;
this.call_state = call_state;
this.stream_interactor = stream_interactor;
this.calls = stream_interactor.get_module(Calls.IDENTITY);
var session_info_type = stream_interactor.module_manager.get_module(call.account, Xep.JingleRtp.Module.IDENTITY).session_info_type;
session_info_type.mute_update_received.connect((session,mute, name) => {
if (this.sid != session.sid) return;
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(mute, rtp_content_parameter.media);
}
}
}
});
session_info_type.info_received.connect((session, session_info) => {
if (this.sid != session.sid) return;
info_received(session_info);
});
}
public async void initiate_call(Jid counterpart) {
Gee.List<Jid> call_resources = yield calls.get_call_resources(call.account, counterpart);
bool do_jmi = false;
Jid? jid_for_direct = null;
if (yield calls.contains_jmi_resources(call.account, call_resources)) {
do_jmi = true;
} else if (!call_resources.is_empty) {
jid_for_direct = call_resources[0];
} else if (calls.has_jmi_resources(jid)) {
do_jmi = true;
}
sid = Xmpp.random_uuid();
if (do_jmi) {
XmppStream? stream = stream_interactor.get_stream(call.account);
var descriptions = new ArrayList<StanzaNode>();
descriptions.add(new StanzaNode.build("description", Xep.JingleRtp.NS_URI).add_self_xmlns().put_attribute("media", "audio"));
if (we_should_send_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, jid, sid, descriptions);
// call_state.cim_invite_id = stream.get_module(Xmpp.Xep.CallInvites.Module.IDENTITY).send_jingle_propose(stream, jid, sid, we_should_send_video);
} else if (jid_for_direct != null) {
yield call_resource(jid_for_direct);
}
}
public async void call_resource(Jid full_jid) {
XmppStream? stream = stream_interactor.get_stream(call.account);
if (stream == null) return;
if (sid == null) sid = Xmpp.random_uuid();
Xep.Jingle.Session session = yield stream.get_module(Xep.JingleRtp.Module.IDENTITY).start_call(stream, full_jid, we_should_send_video, sid, group_call != null ? group_call.muc_jid : null);
set_session(session);
}
public void accept() {
if (session != null) {
foreach (Xep.Jingle.Content content in session.contents) {
content.accept();
}
} else {
// Only a JMI so far
XmppStream stream = stream_interactor.get_stream(call.account);
if (stream == null) return;
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, jid, sid);
}
}
public void reject() {
if (session != null) {
foreach (Xep.Jingle.Content content in session.contents) {
content.reject();
}
} else {
// Only a JMI so far
XmppStream stream = stream_interactor.get_stream(call.account);
if (stream == null) return;
stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_reject_to_peer(stream, jid, sid);
stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_reject_to_self(stream, sid);
}
}
public void end(string terminate_reason, string? reason_text = null) {
switch (terminate_reason) {
case Xep.Jingle.ReasonElement.SUCCESS:
if (session != null) {
session.terminate(terminate_reason, reason_text, "success");
}
break;
case Xep.Jingle.ReasonElement.CANCEL:
if (session != null) {
session.terminate(terminate_reason, reason_text, "cancel");
} else if (group_call != null) {
// We don't have to do anything (?)
} else {
// Only a JMI so far
XmppStream? stream = stream_interactor.get_stream(call.account);
if (stream == null) return;
stream.get_module(Xep.JingleMessageInitiation.Module.IDENTITY).send_session_retract_to_peer(stream, jid, sid);
}
break;
}
}
internal void mute_own_audio(bool mute) {
// Call isn't fully established yet. Audio will be muted once the stream is created.
if (session == null || audio_content_parameter == null || audio_content_parameter.stream == null) return;
Xep.JingleRtp.Stream stream = audio_content_parameter.stream;
// 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(session, mute, "audio");
// Start/Stop sending audio data
Application.get_default().plugin_registry.video_call_plugin.set_pause(stream, mute);
}
internal void mute_own_video(bool mute) {
if (session == null) {
// 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 != null &&
video_content_parameter.stream != null &&
session.senders_include_us(video_content.senders)) {
// A video content already exists
// Start/Stop sending video data
Xep.JingleRtp.Stream stream = video_content_parameter.stream;
if (stream != null) {
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(session, mute, "video");
} else if (!mute) {
// Add a new video content
XmppStream stream = stream_interactor.get_stream(call.account);
rtp_module.add_outgoing_video_content.begin(stream, session, group_call != null ? group_call.muc_jid : null, (_, res) => {
if (video_content_parameter == 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(content, rtp_content_parameter);
}
}
});
}
// If video_content_parameter == null && !mute we're trying to mute a non-existant feed. It will be muted as soon as it is created.
}
public Xep.JingleRtp.Stream? get_video_stream() {
if (video_content_parameter != null) {
return video_content_parameter.stream;
}
return null;
}
public Xep.JingleRtp.Stream? get_audio_stream() {
if (audio_content_parameter != null) {
return audio_content_parameter.stream;
}
return null;
}
internal void set_session(Xep.Jingle.Session session) {
this.session = session;
this.sid = session.sid;
session.terminated.connect((stream, we_terminated, reason_name, reason_text) =>
session_terminated(we_terminated, reason_name, reason_text)
);
session.additional_content_add_incoming.connect((stream, content) =>
on_incoming_content_add(stream, content.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(content, rtp_content_parameter);
}
}
public PeerInfo get_info() {
var ret = new PeerInfo();
if (audio_content != null || audio_content_parameter != null) {
ret.audio = get_content_info(audio_content, audio_content_parameter);
}
if (video_content != null || video_content_parameter != null) {
ret.video = get_content_info(video_content, video_content_parameter);
}
return ret;
}
private PeerContentInfo get_content_info(Xep.Jingle.Content? content, Xep.JingleRtp.Parameters? parameter) {
PeerContentInfo ret = new PeerContentInfo();
if (parameter != null) {
ret.rtcp_ready = parameter.rtcp_ready;
ret.rtp_ready = parameter.rtp_ready;
if (parameter.agreed_payload_type != null) {
ret.codec = parameter.agreed_payload_type.name;
ret.clockrate = parameter.agreed_payload_type.clockrate;
}
if (parameter.stream != null && parameter.stream.remb_enabled) {
ret.target_receive_bytes = parameter.stream.target_receive_bitrate;
ret.target_send_bytes = parameter.stream.target_send_bitrate;
}
}
if (content != null) {
Xmpp.Xep.Jingle.ComponentConnection? component0 = content.get_transport_connection(1);
if (component0 != null) {
ret.bytes_received = component0.bytes_received;
ret.bytes_sent = component0.bytes_sent;
}
}
return ret;
}
private void connect_content_signals(Xep.Jingle.Content content, Xep.JingleRtp.Parameters rtp_content_parameter) {
if (rtp_content_parameter.media == "audio") {
audio_content = content;
audio_content_parameter = rtp_content_parameter;
} else if (rtp_content_parameter.media == "video") {
video_content = content;
video_content_parameter = rtp_content_parameter;
}
debug(@"[%s] %s connecting content signals %s", call.account.bare_jid.to_string(), jid.to_string(), rtp_content_parameter.media);
rtp_content_parameter.stream_created.connect((stream) => on_stream_created(rtp_content_parameter.media, stream));
rtp_content_parameter.connection_ready.connect((status) => {
Idle.add(() => {
on_connection_ready(content, rtp_content_parameter.media);
return false;
});
});
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(false, "video");
}
});
}
private void on_incoming_content_add(XmppStream stream, 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(content, rtp_content_parameter);
content.accept();
}
private void on_stream_created(string media, Xep.JingleRtp.Stream stream) {
if (media == "video" && stream.receiving) {
counterpart_sends_video = true;
video_content_parameter.connection_ready.connect((status) => {
Idle.add(() => {
counterpart_sends_video_updated(false);
return false;
});
});
}
// Outgoing audio/video might have been muted in the meanwhile.
if (media == "video" && !we_should_send_video) {
mute_own_video(true);
} else if (media == "audio" && !we_should_send_audio) {
mute_own_audio(true);
}
stream_created(media);
}
private void on_counterpart_mute_update(bool mute, string? media) {
if (!call.equals(call)) return;
if (media == "video") {
counterpart_sends_video = !mute;
debug(@"[%s] %s video muted %s", call.account.bare_jid.to_string(), jid.to_string(), mute.to_string());
counterpart_sends_video_updated(mute);
}
}
private void on_connection_ready(Xep.Jingle.Content content, string media) {
debug("[%s] %s on_connection_ready", call.account.bare_jid.to_string(), jid.to_string());
connection_ready();
if (call.state == Call.State.RINGING || call.state == Call.State.ESTABLISHING) {
call.state = Call.State.IN_PROGRESS;
}
if (media == "audio") {
audio_encryptions = content.encryptions;
} else if (media == "video") {
video_encryptions = content.encryptions;
}
if ((audio_encryptions != null && audio_encryptions.is_empty) || (video_encryptions != null && video_encryptions.is_empty)) {
call.encryption = Encryption.NONE;
encryption_updated(null, null, true);
return;
}
HashMap<string, Xep.Jingle.ContentEncryption> encryptions = audio_encryptions ?? video_encryptions;
Xep.Jingle.ContentEncryption? omemo_encryption = null, dtls_encryption = null, srtp_encryption = null;
foreach (string encr_name in encryptions.keys) {
if (video_encryptions != null && !video_encryptions.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;
omemo_encryption.peer_key = dtls_encryption.peer_key;
omemo_encryption.our_key = dtls_encryption.our_key;
audio_encryption = omemo_encryption;
encryption_keys_same = true;
video_encryption = video_encryptions != null ? video_encryptions["http://gultsch.de/xmpp/drafts/omemo/dlts-srtp-verification"] : null;
} else if (dtls_encryption != null) {
call.encryption = Encryption.DTLS_SRTP;
audio_encryption = dtls_encryption;
video_encryption = video_encryptions != null ? video_encryptions[Xep.JingleIceUdp.DTLS_NS_URI] : null;
encryption_keys_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]) {
encryption_keys_same = false;
break;
}
}
}
} else if (srtp_encryption != null) {
call.encryption = Encryption.SRTP;
audio_encryption = srtp_encryption;
video_encryption = video_encryptions != null ? video_encryptions["SRTP"] : null;
encryption_keys_same = false;
} else {
call.encryption = Encryption.NONE;
encryption_keys_same = true;
}
encryption_updated(audio_encryption, video_encryption, encryption_keys_same);
}
}
public class Dino.PeerContentInfo {
public bool rtp_ready { get; set; }
public bool rtcp_ready { get; set; }
public ulong? bytes_sent { get; set; default=0; }
public ulong? bytes_received { get; set; default=0; }
public string? codec { get; set; }
public uint32 clockrate { get; set; }
public uint target_receive_bytes { get; set; default=-1; }
public uint target_send_bytes { get; set; default=-1; }
}
public class Dino.PeerInfo {
public PeerContentInfo? audio = null;
public PeerContentInfo? video = null;
}
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