diff options
Diffstat (limited to 'protocols/JabberG/src/jabber_voip.cpp')
| -rw-r--r-- | protocols/JabberG/src/jabber_voip.cpp | 594 |
1 files changed, 0 insertions, 594 deletions
diff --git a/protocols/JabberG/src/jabber_voip.cpp b/protocols/JabberG/src/jabber_voip.cpp deleted file mode 100644 index 9c8cb589d9..0000000000 --- a/protocols/JabberG/src/jabber_voip.cpp +++ /dev/null @@ -1,594 +0,0 @@ -#include "stdafx.h" - -#include <m_voiceservice.h> - -#include <gst/gst.h> -#include <gst/sdp/sdp.h> -#include <gst/rtp/rtp.h> - -#define GST_USE_UNSTABLE_API -#include <gst/webrtc/webrtc.h> - -#pragma comment(lib, "glib-2.0.lib") -#pragma comment(lib, "gobject-2.0.lib") -#pragma comment(lib, "gstreamer-1.0.lib") -#pragma comment(lib, "gstrtp-1.0.lib") -#pragma comment(lib, "gstsdp-1.0.lib") -#pragma comment(lib, "gstwebrtc-1.0.lib") - -static std::list<CMStringA> remotecands; - -bool GetCandidateProp(char *output, byte maxlen, const char *candidate, const char *prop) -{ - const char *pprop = strstr(candidate, prop); - if (!pprop) - return false; - - const char *val = pprop + strlen(prop); - while (*val == ' ') val++; - int i = 0; - while (*val != 0 && *val != ' ' && i < maxlen - 1) - output[i++] = *val++; - output[i] = 0; - - return i > 0; -} - -static void handle_media_stream(GstPad *pad, GstElement *pipe, const char *convert_name, const char *sink_name) -{ - GstPad *qpad; - GstElement *q, *conv, *resample, *sink; - GstPadLinkReturn ret; - - gst_print("Trying to handle stream with %s ! %s", convert_name, sink_name); - - q = gst_element_factory_make("queue", NULL); - g_assert_nonnull(q); - conv = gst_element_factory_make(convert_name, NULL); - g_assert_nonnull(conv); - sink = gst_element_factory_make(sink_name, NULL); - g_assert_nonnull(sink); - - if (g_strcmp0(convert_name, "audioconvert") == 0) { - /* Might also need to resample, so add it just in case. - * Will be a no-op if it's not required. */ - resample = gst_element_factory_make("audioresample", NULL); - g_assert_nonnull(resample); - gst_bin_add_many(GST_BIN(pipe), q, conv, resample, sink, NULL); - gst_element_sync_state_with_parent(q); - gst_element_sync_state_with_parent(conv); - gst_element_sync_state_with_parent(resample); - gst_element_sync_state_with_parent(sink); - gst_element_link_many(q, conv, resample, sink, NULL); - } - else { - gst_bin_add_many(GST_BIN(pipe), q, conv, sink, NULL); - gst_element_sync_state_with_parent(q); - gst_element_sync_state_with_parent(conv); - gst_element_sync_state_with_parent(sink); - gst_element_link_many(q, conv, sink, NULL); - } - - qpad = gst_element_get_static_pad(q, "sink"); - - ret = gst_pad_link(pad, qpad); - g_assert_cmphex(ret, == , GST_PAD_LINK_OK); -} - -static void on_incoming_decodebin_stream(GstElement * /*decodebin*/, GstPad *pad, GstElement *pipe) -{ - GstCaps *caps; - const gchar *name; - - if (!gst_pad_has_current_caps(pad)) { - gst_printerr("Pad '%s' has no caps, can't do anything, ignoring\n", GST_PAD_NAME(pad)); - return; - } - - caps = gst_pad_get_current_caps(pad); - name = gst_structure_get_name(gst_caps_get_structure(caps, 0)); - - if (g_str_has_prefix(name, "video")) { - handle_media_stream(pad, pipe, "videoconvert", "autovideosink"); - } - else if (g_str_has_prefix(name, "audio")) { - handle_media_stream(pad, pipe, "audioconvert", "autoaudiosink"); - } - else { - gst_printerr("Unknown pad %s, ignoring", GST_PAD_NAME(pad)); - } -} - -static void on_incoming_stream_cb(GstElement */*webrtc*/, GstPad *pad, GstElement *pipe) -{ - GstElement *decodebin; - GstPad *sinkpad; - - if (GST_PAD_DIRECTION(pad) != GST_PAD_SRC) - return; - - decodebin = gst_element_factory_make("decodebin", NULL); - g_signal_connect(decodebin, "pad-added", G_CALLBACK(on_incoming_decodebin_stream), pipe); - gst_bin_add(GST_BIN(pipe), decodebin); - gst_element_sync_state_with_parent(decodebin); - - sinkpad = gst_element_get_static_pad(decodebin, "sink"); - gst_pad_link(pad, sinkpad); - gst_object_unref(sinkpad); -} - -void on_offer_created_cb(GstPromise *promise, gpointer user_data) -{ - GstWebRTCSessionDescription *offer = NULL; - CJabberProto *jproto = (CJabberProto *)user_data; - - GstStructure const *reply = gst_promise_get_reply(promise); - gst_structure_get(reply, jproto->m_isOutgoing ? "offer" : "answer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL); - gst_promise_unref(promise); - if (!offer) { - gst_print("Cannot process sdp"); - return; - } - - GstPromise *local_desc_promise = gst_promise_new(); - g_signal_emit_by_name(jproto->m_webrtc1, "set-local-description", offer, local_desc_promise); - gst_promise_interrupt(local_desc_promise); - gst_promise_unref(local_desc_promise); - - gchar *sdp_string = gst_sdp_message_as_text(offer->sdp); - gst_print("VOIP - Wanna send SDP offer:\r\n%s\r\n", sdp_string); - g_free(sdp_string); - - const GstSDPMedia *media_audio = NULL; - for (unsigned int i = 0; i < gst_sdp_message_medias_len(offer->sdp); i++) { - const GstSDPMedia *m = gst_sdp_message_get_media(offer->sdp, i); - if (!strcmp(m->media, "audio")) - media_audio = m; - } - if (!media_audio) { - gst_print("No audio media in SDP"); - return; - } - - jproto->m_voipICEPwd = gst_sdp_media_get_attribute_val(media_audio, "ice-pwd"); - jproto->m_voipICEUfrag = gst_sdp_media_get_attribute_val(media_audio, "ice-ufrag"); - jproto->m_medianame = gst_sdp_media_get_attribute_val(media_audio, "mid"); - - //send it all - bool outgoing = jproto->m_isOutgoing; - XmlNodeIq iq("set", jproto->SerialNext(), jproto->m_voipPeerJid); - TiXmlElement *rjNode = iq << XCHILDNS("jingle", JABBER_FEAT_JINGLE); - rjNode << XATTR("sid", jproto->m_voipSession) - << XATTR("action", outgoing ? "session-initiate" : "session-accept") - << XATTR("initiator", outgoing ? jproto->m_ThreadInfo->fullJID : jproto->m_voipPeerJid); - if (!outgoing) - rjNode << XATTR("responder", jproto->m_ThreadInfo->fullJID); - - TiXmlElement *content = rjNode << XCHILD("content") << XATTR("creator", "initiator") << XATTR("name", jproto->m_medianame); - TiXmlElement *description = content << XCHILDNS("description", JABBER_FEAT_JINGLE_RTP) << XATTR("media", "audio"); - - auto *opuspayload = description << XCHILD("payload-type") << XATTR("id", "111") << XATTR("name", "opus") << XATTR("clockrate", "48000") << XATTR("channels", "2"); - - opuspayload << XCHILD("parameter") << XATTR("name", "minptime") << XATTR("value", "10"); - opuspayload << XCHILD("parameter") << XATTR("name", "useinbandfec") << XATTR("value", "1"); - opuspayload << XCHILDNS("rtcp-fb", "urn:xmpp:jingle:apps:rtp:rtcp-fb:0") << XATTR("type", "transport-cc"); - - description << XCHILDNS("rtp-hdrext", "urn:xmpp:jingle:apps:rtp:rtp-hdrext:0") << XATTR("id", "1") << XATTR("uri", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"); - /* - auto* source = description << XCHILDNS("source", "urn:xmpp:jingle:apps:rtp:ssma:0") << XATTR("ssrc", "2165039095"); - source << XCHILD("parameter") << XATTR("name", "cname") << XATTR("value", "8ee+PcGu8BNwq22f"); - source << XCHILD("parameter") << XATTR("name", "msid") << XATTR("value", "my-media-stream2 my-audio-track2"); - source << XCHILD("parameter") << XATTR("name", "mslabel") << XATTR("value", "my-media-stream2"); - source << XCHILD("parameter") << XATTR("name", "label") << XATTR("value", "my-audio-track2");*/ - - description << XCHILD("rtcp-mux"); - - //fingerprint - char hash[100]; - if (sscanf(gst_sdp_media_get_attribute_val(media_audio, "fingerprint"), "sha-256 %95s", hash) == 1) { - auto *transport = content << XCHILDNS("transport", JABBER_FEAT_JINGLE_ICEUDP); - transport << XATTR("pwd", jproto->m_voipICEPwd) << XATTR("ufrag", jproto->m_voipICEUfrag); - - auto *fingerprint = transport << XCHILD("fingerprint", hash); - fingerprint << XATTR("xmlns", JABBER_FEAT_JINGLE_DTLS) << XATTR("hash", "sha-256") - << XATTR("setup", gst_sdp_media_get_attribute_val(media_audio, "setup")); - } - - jproto->m_ThreadInfo->send(iq); - - gst_webrtc_session_description_free(offer); -} - -void on_negotiation_needed_cb(GstElement *webrtcbin, gpointer user_data) -{ - if (((CJabberProto *)user_data)->m_isOutgoing) { - gst_print("Creating negotiation offer\n"); - - GstPromise *promise = gst_promise_new_with_change_func(on_offer_created_cb, user_data, NULL); - g_signal_emit_by_name(G_OBJECT(webrtcbin), "create-offer", NULL, promise); - } -} - -static void on_offer_set(GstPromise *promise, gpointer user_data) -{ - gst_promise_unref(promise); - promise = gst_promise_new_with_change_func(on_offer_created_cb, user_data, NULL); - g_signal_emit_by_name(((CJabberProto *)user_data)->m_webrtc1, "create-answer", NULL, promise); -} - -void send_ice_candidate_message_cb(G_GNUC_UNUSED GstElement */*webrtcbin*/, guint mline_index, gchar *candidate, CJabberProto *jproto) -{ - // parse candidate and send - char foundation[11], component[11], protocol[4], priority[11], ip[40], port[6], type[6]; - int ret = sscanf(candidate, "candidate:%10s %10s %3s %10s %39s %5s typ %5s", - foundation, component, protocol, priority, ip, port, type); - if (ret != 7 || strcmp(protocol, "UDP")) - return; - - gst_print("VOIP - Wanna send ice candidate(m-line_index=%d):\r\n%s\r\n", mline_index, candidate); - for (char *p = protocol; *p; ++p) *p = tolower(*p); - - XmlNodeIq iq("set", jproto->SerialNext(), jproto->m_voipPeerJid); - TiXmlElement *rjNode = iq << XCHILDNS("jingle", JABBER_FEAT_JINGLE); - rjNode << XATTR("action", "transport-info") << XATTR("sid", jproto->m_voipSession); - - TiXmlElement *content = rjNode << XCHILD("content"); - content << XATTR("creator", "initiator") << XATTR("name", jproto->m_medianame); - - auto *transport = content << XCHILDNS("transport", JABBER_FEAT_JINGLE_ICEUDP); - transport << XATTR("pwd", jproto->m_voipICEPwd) << XATTR("ufrag", jproto->m_voipICEUfrag); - - auto *candidateNode = transport << XCHILD("candidate"); - candidateNode << XATTR("type", type) << XATTR("protocol", protocol) << XATTR("ip", ip) - << XATTR("port", port) << XATTR("priority", priority) << XATTR("foundation", foundation) << XATTR("component", component); - - char attr[255]; - if (GetCandidateProp(attr, 255, candidate, "raddr")) - candidateNode << XATTR("rel-addr", attr); - if (GetCandidateProp(attr, 255, candidate, "rport")) - candidateNode << XATTR("rel-port", attr); - - jproto->m_ThreadInfo->send(iq); -} - -static gboolean check_plugins(void) -{ - const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp", "rtpmanager" - /*"vpx", "videotestsrc", "audiotestsrc",*/ }; - - GstRegistry *registry = gst_registry_get(); - gboolean ret = TRUE; - for (auto &it : needed) { - GstPlugin *plugin = gst_registry_find_plugin(registry, it); - if (!plugin) { - gst_print("Required gstreamer plugin '%s' not found\n", it); - ret = FALSE; - } - else gst_object_unref(plugin); - } - - return ret; -} - -void dbgprint(const gchar *string) -{ - OutputDebugStringA(string); -} - -bool CJabberProto::VOIPCreatePipeline(void) -{ - if (!hasJingle()) - goto err; - - //gstreamer init - static bool gstinited = 0; - if (!gstinited) { - if (!LoadLibrary(L"gstreamer-1.0-0.dll")) { - MessageBoxA(0, "Cannot load Gstreamer library!", 0, MB_OK | MB_ICONERROR); - goto err; - } - gst_init(NULL, NULL); - g_set_print_handler(dbgprint); - gst_print("preved medved"); - if (!check_plugins()) { - MessageBoxA(0, "Gstreamer plugins not found!", 0, MB_OK | MB_ICONERROR); - goto err; - } - gstinited = 1; - } - - #define STUN_SERVER "stun-server=stun://stun.tng.de:3478 " - #define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload=" - #define RTP_TWCC_URI "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01" - - GError *error = NULL; - m_pipe1 = gst_parse_launch( - "webrtcbin bundle-policy=max-bundle name=sendrecv " - STUN_SERVER - "autoaudiosrc ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! " - "queue ! " RTP_CAPS_OPUS "111 ! sendrecv. ", &error); - - if (error) { - MessageBoxA(0, "Failed to parse launch: ", error->message, MB_OK); - g_error_free(error); - goto err; - } - - m_webrtc1 = gst_bin_get_by_name(GST_BIN(m_pipe1), "sendrecv"); - g_assert_nonnull(m_webrtc1); - if (!m_webrtc1) - goto err; - - GstElement *audiopay = gst_bin_get_by_name(GST_BIN(m_pipe1), "audiopay"); - g_assert_nonnull(audiopay); - GstRTPHeaderExtension *audio_twcc = gst_rtp_header_extension_create_from_uri(RTP_TWCC_URI); - g_assert_nonnull(audio_twcc); - gst_rtp_header_extension_set_id(audio_twcc, 1); - g_signal_emit_by_name(audiopay, "add-extension", audio_twcc); - g_clear_object(&audio_twcc); - g_clear_object(&audiopay); - - // It will be called when the pipeline goes to PLAYING. - g_signal_connect(m_webrtc1, "on-negotiation-needed", G_CALLBACK(on_negotiation_needed_cb), this); - // It will be called when we obtain local ICE candidate - g_signal_connect(m_webrtc1, "on-ice-candidate", G_CALLBACK(send_ice_candidate_message_cb), this); - // idk - g_signal_connect(m_webrtc1, "pad-added", G_CALLBACK(on_incoming_stream_cb), m_pipe1); - - // Lifetime is the same as the pipeline itself - gst_object_unref(m_webrtc1); - - gst_print("Starting pipeline\n"); - if (gst_element_set_state(GST_ELEMENT(m_pipe1), GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE) - return true; - -err: - VOIPTerminateSession(); - return false; -} - -bool CJabberProto::VOIPTerminateSession(const char *reason) -{ - if (m_pipe1) { - gst_element_set_state(GST_ELEMENT(m_pipe1), GST_STATE_NULL); - g_clear_object(&m_pipe1); - gst_object_unref(m_pipe1); - gst_print("Pipeline stopped\n"); - } - - if (m_ThreadInfo && reason && !m_voipSession.IsEmpty() && !m_voipPeerJid.IsEmpty()) { - XmlNodeIq iq("set", SerialNext(), m_voipPeerJid); - - TiXmlElement *jingleNode = iq << XCHILDNS("jingle", JABBER_FEAT_JINGLE); - jingleNode << XATTR("action", "session-terminate") << XATTR("sid", m_voipSession); - jingleNode << XATTR("initiator", m_isOutgoing ? m_ThreadInfo->fullJID : m_voipPeerJid); - jingleNode << XCHILD("reason") << XCHILD(reason); - - m_ThreadInfo->send(iq); - } - - m_voipICEPwd.Empty(); - m_voipICEUfrag.Empty(); - m_medianame.Empty(); - - m_voipSession.Empty(); - m_voipPeerJid.Empty(); - m_pipe1 = m_webrtc1 = NULL; - return true; -} - -bool CJabberProto::OnRTPDescription(const TiXmlElement *jingleNode) -{ - if (!jingleNode) - return false; - - // process remote offer - auto *content = XmlGetChildByTag(jingleNode, "content", "creator", "initiator"); - auto *transport = XmlGetChildByTag(content, "transport", "xmlns", "urn:xmpp:jingle:transports:ice-udp:1"); - auto *description = XmlGetChildByTag(content, "description", "xmlns", "urn:xmpp:jingle:apps:rtp:1"); - auto *source = XmlGetChildByTag(description, "source", "xmlns", "urn:xmpp:jingle:apps:rtp:ssma:0"); - - CMStringA sdp_string(FORMAT, "v=0\r\no=- 0 0 IN IP4 0.0.0.0\r\ns=-\r\nt=0 0\r\na=ice-options:trickle\r\n" - "m=audio 9 UDP/TLS/RTP/SAVPF 111\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:%s\r\na=ice-pwd:%s\r\na=rtcp-mux\r\na=sendrecv\r\na=rtpmap:111 OPUS/48000/2\r\n" - - "a=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\n" - "a=ssrc:%s msid:%s\r\n" - "a=ssrc:%s cname:%s\r\n" - - "a=mid:%s\r\na=setup:%s\r\na=fingerprint:sha-256 %s\r\na=rtcp-mux-only\r\n", - XmlGetAttr(transport, "ufrag"), - XmlGetAttr(transport, "pwd"), - - XmlGetAttr(source, "ssrc"), - XmlGetAttr(XmlGetChildByTag(source, "parameter", "name", "msid"), "value"), - XmlGetAttr(source, "ssrc"), - XmlGetAttr(XmlGetChildByTag(source, "parameter", "name", "cname"), "value"), - - XmlGetAttr(content, "name"), - XmlGetAttr(XmlFirstChild(transport, "fingerprint"), "setup"), - XmlFirstChild(transport, "fingerprint")->GetText()); - - GstSDPMessage *sdp; - int ret = gst_sdp_message_new(&sdp); - g_assert_cmphex(ret, == , GST_SDP_OK); - ret = gst_sdp_message_parse_buffer((guint8 *)sdp_string.c_str(), sdp_string.GetLength(), sdp); - if (ret != GST_SDP_OK) { - g_error("Could not parse SDP string\n"); - return false; - } - - gchar *str = gst_sdp_message_as_text(sdp); - gst_print("VOIP - Eating remote SDP offer:\r\n%s\r\n", str); - g_free(str); - - if (m_isOutgoing) { - GstWebRTCSessionDescription *answer = gst_webrtc_session_description_new(GST_WEBRTC_SDP_TYPE_ANSWER, sdp); - g_assert_nonnull(answer); - - GstPromise *promise = gst_promise_new(); - g_signal_emit_by_name(m_webrtc1, "set-remote-description", answer, promise); - gst_promise_interrupt(promise); - gst_promise_unref(promise); - gst_webrtc_session_description_free(answer); - } - else { - // Set remote description on our pipeline - GstWebRTCSessionDescription *offer = gst_webrtc_session_description_new(GST_WEBRTC_SDP_TYPE_OFFER, sdp); - g_assert_nonnull(offer); - - GstPromise *promise = gst_promise_new_with_change_func(on_offer_set, this, NULL); - g_signal_emit_by_name(m_webrtc1, "set-remote-description", offer, promise); - gst_webrtc_session_description_free(offer); - } - - return true; -} - -bool CJabberProto::OnICECandidate(const TiXmlElement *Node) -{ - if (!hasJingle()) - return false; - - CMStringA scandidate; - CMStringA proto(XmlGetAttr(Node, "protocol")); - proto.MakeUpper(); - - scandidate.AppendFormat("candidate:%s ", XmlGetAttr(Node, "foundation")); //FIXME - scandidate.AppendFormat("%s ", XmlGetAttr(Node, "component")); - scandidate.AppendFormat("%s ", proto.c_str()); - scandidate.AppendFormat("%s ", XmlGetAttr(Node, "priority")); - scandidate.AppendFormat("%s ", XmlGetAttr(Node, "ip")); - scandidate.AppendFormat("%s ", XmlGetAttr(Node, "port")); - scandidate.AppendFormat("typ %s", XmlGetAttr(Node, "type")); - - if (const char *tmp = XmlGetAttr(Node, "rel-addr")) - scandidate.AppendFormat(" raddr %s", tmp); - if (const char *tmp = XmlGetAttr(Node, "rel-port")) - scandidate.AppendFormat(" rport %s", tmp); - if (const char *generation = XmlGetAttr(Node, "generation")) - scandidate.AppendFormat(" generation %s", generation); - - gst_print("VOIP - Accepting ICE candidate:\r\n%s\r\n", scandidate.c_str()); - g_signal_emit_by_name(m_webrtc1, "add-ice-candidate", 0, scandidate.c_str()); - return true; -} - -bool CJabberProto::VOIPCallIinitiate(MCONTACT hContact) -{ - if (!m_voipSession.IsEmpty()) { - VOIPTerminateSession(); - MessageBoxA(0, "Something went wrong\r\nOld session terminated", NULL, 0); - return false; - } - - if (!hasJingle()) - return false; - - CMStringA jid(getMStringA(hContact, "jid")); - if (jid.IsEmpty()) - return false; - - auto r = ListGetBestResource(jid); - if (r) { - bool bFound = false; - if (auto *pFeature = FindFeature(JABBER_FEAT_JINGLE)) - if (!(r->m_pCaps->GetCaps() & pFeature->jcbCap)) - bFound = true; - - if (!bFound) { - MsgPopup(hContact, TranslateT("Client's program does not support voice calls"), TranslateT("Error")); - return false; - } - - jid = MakeJid(jid, r->m_szResourceName); - } - - unsigned char tmp[16]; - Utils_GetRandom(tmp, sizeof(tmp)); - - m_isOutgoing = true; - m_voipSession = ptrA(mir_base64_encode(tmp, sizeof(tmp))); - m_voipPeerJid = jid.c_str(); - - return true; -} - -INT_PTR CJabberProto::JabberVOIP_call(WPARAM hContact, LPARAM) -{ - if (VOIPCallIinitiate(hContact)) { - VOICE_CALL vc = {}; - vc.cbSize = sizeof(VOICE_CALL); - vc.moduleName = m_szModuleName; - vc.id = m_voipSession; // Protocol especific ID for this call - vc.hContact = hContact; // Contact associated with the call (can be NULL) - vc.state = VOICE_STATE_READY; - vc.szNumber.a = m_voipPeerJid; - NotifyEventHooks(m_hVoiceEvent, WPARAM(&vc), 0); - } - - return 0; -} - -INT_PTR CJabberProto::JabberVOIP_answercall(WPARAM id, LPARAM) -{ - if(strcmp((const char *)id, m_voipSession)) - return 0; - -/* CMStringA question(FORMAT, "Proceed call with %s?\r\n" - "It will disclose IP address to the peer and his server", m_voipPeerJid.c_str()); - if (MessageBoxA(0, question.c_str(), "Outgoing call", MB_YESNO | MB_ICONQUESTION) != IDYES) - return 0;*/ - - VOICE_CALL vc = {}; - vc.cbSize = sizeof(VOICE_CALL); - vc.moduleName = m_szModuleName; - vc.hContact = HContactFromJID(m_voipPeerJid);// Contact associated with the call (can be NULL) - vc.szNumber.a = m_voipPeerJid; - vc.id = m_voipSession; - vc.state = VOICE_STATE_ENDED; - - if (VOIPCreatePipeline()) { - if (m_isOutgoing) - vc.state = VOICE_STATE_CALLING; - else if (OnRTPDescription(m_offerNode)) - vc.state = VOICE_STATE_TALKING; - else - VOIPTerminateSession(); - } - NotifyEventHooks(m_hVoiceEvent, WPARAM(&vc), 0); - - return 0; -} - -INT_PTR CJabberProto::JabberVOIP_dropcall(WPARAM id, LPARAM) -{ - VOICE_CALL vc = {}; - vc.cbSize = sizeof(VOICE_CALL); - vc.moduleName = m_szModuleName; - vc.id = (char*)id; - vc.state = VOICE_STATE_ENDED; - NotifyEventHooks(m_hVoiceEvent, WPARAM(&vc), 0); - - VOIPTerminateSession(); - return 0; -} - -///////////////////////////////////////////////////////////////////////////////////////// -// module entry point - -void CJabberProto::InitVoip(bool bEnable) -{ - // Voip - VOICE_MODULE vsr = {}; - vsr.cbSize = sizeof(VOICE_MODULE); - vsr.description = L"XMPP/DTLS-SRTP"; - vsr.name = m_szModuleName; - vsr.icon = g_plugin.getIconHandle(IDI_NOTES); - vsr.flags = 3; - if (bEnable) - CallService(MS_VOICESERVICE_REGISTER, (WPARAM)&vsr, 0); - else { - VOIPTerminateSession(); - CallService(MS_VOICESERVICE_UNREGISTER, (WPARAM)&vsr, 0); - } -} |