Jabber: initial version of Jingle support

10 files changed, 649 insertions, 33 deletions

diff --git a/protocols/JabberG/jabber.vcxproj b/protocols/JabberG/jabber.vcxproj
index 4d882b075d..cffe651bac 100644
--- a/protocols/JabberG/jabber.vcxproj
+++ b/protocols/JabberG/jabber.vcxproj
@@ -25,6 +25,11 @@
   <ImportGroup Label="PropertySheets">

     <Import Project="$(ProjectDir)..\..\build\vc.common\plugin.props" />

   </ImportGroup>

+  <ItemDefinitionGroup>

+    <ClCompile>

+      <AdditionalIncludeDirectories>..\..\include\glib;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+    </ClCompile>

+  </ItemDefinitionGroup>

   <ItemGroup>

     <ClCompile Include="src\jabber.cpp" />

     <ClCompile Include="src\jabber_adhoc.cpp" />

@@ -75,6 +80,7 @@
     <ClCompile Include="src\jabber_userinfo.cpp" />

     <ClCompile Include="src\jabber_util.cpp" />

     <ClCompile Include="src\jabber_vcard.cpp" />

+    <ClCompile Include="src\jabber_voip.cpp" />

     <ClCompile Include="src\jabber_xml.cpp" />

     <ClCompile Include="src\jabber_xstatus.cpp" />

     <ClCompile Include="src\jabber_zstream.cpp" />

@@ -107,7 +113,8 @@
   </ItemGroup>

   <ItemDefinitionGroup>

     <Link>

-      <AdditionalDependencies>libcrypto.lib;libssl.lib;%(AdditionalDependencies)</AdditionalDependencies>

+      <AdditionalDependencies>delayimp.lib;libcrypto.lib;libssl.lib;%(AdditionalDependencies)</AdditionalDependencies>

+      <DelayLoadDLLs>glib-2.0-0.dll;gobject-2.0-0.dll;gstreamer-1.0-0.dll;gstrtp-1.0-0.dll;gstsdp-1.0-0.dll;gstwebrtc-1.0-0.dll;%(DelayLoadDLLs)</DelayLoadDLLs>

     </Link>

   </ItemDefinitionGroup>

   <ItemGroup>

diff --git a/protocols/JabberG/jabber.vcxproj.filters b/protocols/JabberG/jabber.vcxproj.filters
index 30d426c7d2..7894068683 100644
--- a/protocols/JabberG/jabber.vcxproj.filters
+++ b/protocols/JabberG/jabber.vcxproj.filters
@@ -161,6 +161,9 @@
     <ClCompile Include="src\jabber_mam.cpp">

       <Filter>Source Files</Filter>

     </ClCompile>

+    <ClCompile Include="src\jabber_voip.cpp">

+      <Filter>Source Files</Filter>

+    </ClCompile>

   </ItemGroup>

   <ItemGroup>

     <ClInclude Include="src\jabber_byte.h">

diff --git a/protocols/JabberG/src/jabber_caps.cpp b/protocols/JabberG/src/jabber_caps.cpp
index 4b6fada073..e49f93d8a8 100644
--- a/protocols/JabberG/src/jabber_caps.cpp
+++ b/protocols/JabberG/src/jabber_caps.cpp
@@ -78,7 +78,11 @@ const JabberFeatCapPair g_JabberFeatCapPairs[] =
 	{ JABBER_FEAT_ROSTER_EXCHANGE,         JABBER_CAPS_ROSTER_EXCHANGE,         LPGEN("Supports Roster Exchange") },

 	{ JABBER_FEAT_DIRECT_MUC_INVITE,       JABBER_CAPS_DIRECT_MUC_INVITE,       LPGEN("Supports direct chat invitations (XEP-0249)") },

 	{ JABBER_FEAT_OMEMO_DEVICELIST_NOTIFY, JABBER_CAPS_OMEMO_DEVICELIST_NOTIFY, LPGEN("Receives information about OMEMO devices") },

-	{ JABBER_FEAT_CARBONS,				      JABBER_CAPS_CARBONS,                 LPGEN("Supports message carbons (XEP-0280)")},

+	{ JABBER_FEAT_CARBONS,				   JABBER_CAPS_CARBONS,                 LPGEN("Supports message carbons (XEP-0280)")},

+	{ JABBER_FEAT_JINGLE_ICEUDP,		   JABBER_CAPS_JINGLE_ICEUDP,           LPGEN("Jingle ICE-UDP Transport") },

+	{ JABBER_FEAT_JINGLE_RTP,			   JABBER_CAPS_JINGLE_RTP,              LPGEN("Jingle RTP") },

+	{ JABBER_FEAT_JINGLE_DTLS,			   JABBER_CAPS_JINGLE_DTLS,             LPGEN("Jingle DTLS") },

+	{ JABBER_FEAT_JINGLE_RTPAUDIO, 		   JABBER_CAPS_JINGLE_RTPAUDIO,         LPGEN("Jingle RTP Audio") },

 };

 

 const int g_cJabberFeatCapPairs = _countof(g_JabberFeatCapPairs);

@@ -347,6 +351,8 @@ JabberCapsBits CJabberProto::GetOwnCaps(bool IncludeDynamic)
 		jcb |= JABBER_CAPS_OMEMO_DEVICELIST_NOTIFY;

 	if (!m_bMsgAck)

 		jcb &= ~(JABBER_CAPS_CHAT_MARKERS | JABBER_CAPS_MESSAGE_RECEIPTS);

+	if (m_bEnableVOIP)

+		jcb |= JABBER_CAPS_JINGLE | JABBER_CAPS_JINGLE_ICEUDP | JABBER_CAPS_JINGLE_RTP | JABBER_CAPS_JINGLE_DTLS | JABBER_CAPS_JINGLE_RTPAUDIO;

 

 	return jcb;

 }

diff --git a/protocols/JabberG/src/jabber_caps.h b/protocols/JabberG/src/jabber_caps.h
index 76313d77aa..fc37993fb0 100644
--- a/protocols/JabberG/src/jabber_caps.h
+++ b/protocols/JabberG/src/jabber_caps.h
@@ -195,6 +195,15 @@ typedef unsigned __int64 JabberCapsBits;
 #define JABBER_FEAT_BITS                        "urn:xmpp:bob"

 #define JABBER_CAPS_BITS                        ((JabberCapsBits)1<<50)

 

+#define JABBER_FEAT_JINGLE_ICEUDP               "urn:xmpp:jingle:transports:ice-udp:1"

+#define JABBER_CAPS_JINGLE_ICEUDP               ((JabberCapsBits)1<<51)

+#define JABBER_FEAT_JINGLE_RTP                  "urn:xmpp:jingle:apps:rtp:1"

+#define JABBER_CAPS_JINGLE_RTP                  ((JabberCapsBits)1<<52)

+#define JABBER_FEAT_JINGLE_DTLS                 "urn:xmpp:jingle:apps:dtls:0"

+#define JABBER_CAPS_JINGLE_DTLS                 ((JabberCapsBits)1<<53)

+#define JABBER_FEAT_JINGLE_RTPAUDIO             "urn:xmpp:jingle:apps:rtp:audio"

+#define JABBER_CAPS_JINGLE_RTPAUDIO             ((JabberCapsBits)1<<54)

+

 #define JABBER_FEAT_ARCHIVE                     "urn:xmpp:archive"

 #define JABBER_FEAT_BIND                        "urn:ietf:params:xml:ns:xmpp-bind"

 #define JABBER_FEAT_CAPTCHA                     "urn:xmpp:captcha"

diff --git a/protocols/JabberG/src/jabber_menu.cpp b/protocols/JabberG/src/jabber_menu.cpp
index f37afe8eb4..808302b164 100644
--- a/protocols/JabberG/src/jabber_menu.cpp
+++ b/protocols/JabberG/src/jabber_menu.cpp
@@ -126,6 +126,13 @@ static int JabberPrebuildContactMenu(WPARAM hContact, LPARAM lParam)
 	return(ppro) ? ppro->OnPrebuildContactMenu(hContact, lParam) : 0;

 }

 

+static INT_PTR JabberVOIPCallIinitiate(WPARAM hContact, LPARAM)

+{

+	CJabberProto *ppro = CMPlugin::getInstance(hContact);

+	ppro->VOIPCallIinitiate(hContact);

+	return 0;

+}

+

 void g_MenuInit(void)

 {

 	hStatusMenuInit = CreateHookableEvent(ME_JABBER_MENUINIT);

@@ -245,6 +252,16 @@ void g_MenuInit(void)
 	mi.hIcolibItem = g_plugin.getIconHandle(IDI_NODE_SERVER);

 	g_hMenuResourcesServer = Menu_AddContactMenuItem(&mi);

 	CreateServiceFunctionParam(mi.pszService, JabberMenuHandleResource, MENUITEM_SERVER);

+

+	SET_UID(mi, 0x2fe60fc5, 0x6417, 0x4f37, 0xa0, 0xbe, 0xa0, 0x59, 0x94, 0x11, 0x1d, 0xd9);

+	mi.root = nullptr;

+	mi.flags = 0;

+	mi.hIcolibItem = g_plugin.getIconHandle(IDI_NODE_RSS);

+	mi.name.a = LPGEN("Voice call");

+	mi.pszService = "Jabber/VOIPCallIinitiate";

+	mi.position = -1999902010;

+	Menu_AddContactMenuItem(&mi);

+	CreateServiceFunction(mi.pszService, JabberVOIPCallIinitiate);

 }

 

 void g_MenuUninit(void)

diff --git a/protocols/JabberG/src/jabber_opt.cpp b/protocols/JabberG/src/jabber_opt.cpp
index 00313c3d5d..c23c101555 100644
--- a/protocols/JabberG/src/jabber_opt.cpp
+++ b/protocols/JabberG/src/jabber_opt.cpp
@@ -706,8 +706,9 @@ public:
 		m_options.AddOption(LPGENW("Messaging"), LPGENW("Enable server-side history (XEP-0136)"), m_proto->m_bEnableMsgArchive);

 		m_options.AddOption(LPGENW("Messaging"), LPGENW("Enable Message Archive Management (XEP-0313)"), m_proto->m_bEnableMam);

 		m_options.AddOption(LPGENW("Messaging"), LPGENW("Enable carbon copies (XEP-0280)"), m_proto->m_bEnableCarbons);

-		m_options.AddOption(LPGENW("Messaging"), LPGENW("Use Stream Management (XEP-0198) if possible (Testing)"), m_proto->m_bEnableStreamMgmt);

+		m_options.AddOption(LPGENW("Messaging"), LPGENW("Enable VOIP(for developers)"), m_proto->m_bEnableVOIP);

 

+		m_options.AddOption(LPGENW("Server options"), LPGENW("Use Stream Management (XEP-0198) if possible (Testing)"), m_proto->m_bEnableStreamMgmt);

 		m_options.AddOption(LPGENW("Server options"), LPGENW("Disable SASL authentication (for old servers)"), m_proto->m_bDisable3920auth);

 		m_options.AddOption(LPGENW("Server options"), LPGENW("Enable stream compression (if possible)"), m_proto->m_bEnableZlib);

 

diff --git a/protocols/JabberG/src/jabber_proto.cpp b/protocols/JabberG/src/jabber_proto.cpp
index 946ae533f4..cdae4a4df3 100644
--- a/protocols/JabberG/src/jabber_proto.cpp
+++ b/protocols/JabberG/src/jabber_proto.cpp
@@ -101,6 +101,7 @@ CJabberProto::CJabberProto(const char *aProtoName, const wchar_t *aUserName) :
 	m_bEnableUserActivity(this, "EnableUserActivity", true),

 	m_bEnableUserMood(this, "EnableUserMood", true),

 	m_bEnableUserTune(this, "EnableUserTune", false),

+	m_bEnableVOIP(this, "EnableVOIP", false),

 	m_bEnableZlib(this, "EnableZlib", true),

 	m_bFixIncorrectTimestamps(this, "FixIncorrectTimestamps", true),

 	m_bGcLogAffiliations(this, "GcLogAffiliations", false),

diff --git a/protocols/JabberG/src/jabber_proto.h b/protocols/JabberG/src/jabber_proto.h
index ce8a4c0881..e13cd79e3e 100644
--- a/protocols/JabberG/src/jabber_proto.h
+++ b/protocols/JabberG/src/jabber_proto.h
@@ -203,6 +203,7 @@ struct CJabberProto : public PROTO<CJabberProto>, public IJabberInterface
 	CMOption<bool> m_bEnableUserActivity;

 	CMOption<bool> m_bEnableUserMood;

 	CMOption<bool> m_bEnableUserTune;

+	CMOption<bool> m_bEnableVOIP;

 	CMOption<bool> m_bEnableZlib;

 	CMOption<bool> m_bFixIncorrectTimestamps;

 	CMOption<bool> m_bGcLogAffiliations;

@@ -886,6 +887,21 @@ struct CJabberProto : public PROTO<CJabberProto>, public IJabberInterface
 	void       SetServerVcard(BOOL bPhotoChanged, wchar_t* szPhotoFileName);

 	void       SaveVcardToDB(HWND hwndPage, int iPage);

 

+	//---- jabber_voip.c -----------------------------------------------------------------

+

+	bool OnICECandidate(const TiXmlElement *Node, const char *from);

+	bool OnRTPDescription(const TiXmlElement *Node);

+	bool VOIPCreatePipeline();

+	bool VOIPTerminateSession();

+	bool VOIPCallAccept(const TiXmlElement *jingleNode, const char *from);

+	bool VOIPCallIinitiate(MCONTACT hContact);

+

+	CMStringA m_voipSession, m_voipPeerJid;

+	CMStringA m_voipICEPwd, m_voipICEUfrag, m_medianame;

+	bool m_isOutgoing;

+	struct _GstElement *m_pipe1 = NULL;

+	struct _GstElement *m_webrtc1 = NULL;

+

 	//---- jabber_xml.c ------------------------------------------------------------------

 

 	void       OnConsoleProcessXml(const TiXmlElement *node, uint32_t flags);

diff --git a/protocols/JabberG/src/jabber_thread.cpp b/protocols/JabberG/src/jabber_thread.cpp
index cedb5ed81b..ce3c613013 100644
--- a/protocols/JabberG/src/jabber_thread.cpp
+++ b/protocols/JabberG/src/jabber_thread.cpp
@@ -1770,41 +1770,78 @@ bool CJabberProto::OnProcessJingle(const TiXmlElement *node)
 	if (child) {

 		if ((type = XmlGetAttr(node, "type")) == nullptr)

 			return false;

+		const char *szAction = XmlGetAttr(child, "action");

+		const char *szSid = XmlGetAttr(child, "sid");

 

 		if ((!mir_strcmp(type, "get") || !mir_strcmp(type, "set"))) {

-			const char *szAction = XmlGetAttr(child, "action");

 			const char *idStr = XmlGetAttr(node, "id");

 			const char *from = XmlGetAttr(node, "from");

-			if (szAction && !mir_strcmp(szAction, "session-initiate")) {

-				// if this is a Jingle 'session-initiate' and noone processed it yet, reply with "unsupported-applications"

-				m_ThreadInfo->send(XmlNodeIq("result", idStr, from));

-

-				XmlNodeIq iq("set", SerialNext(), from);

-				TiXmlElement *jingleNode = iq << XCHILDNS("jingle", JABBER_FEAT_JINGLE);

-

-				jingleNode << XATTR("action", "session-terminate");

-				const char *szInitiator = XmlGetAttr(child, "initiator");

-				if (szInitiator)

-					jingleNode << XATTR("initiator", szInitiator);

-				const char *szSid = XmlGetAttr(child, "sid");

-				if (szSid)

-					jingleNode << XATTR("sid", szSid);

-

-				jingleNode << XCHILD("reason")

-					<< XCHILD("unsupported-applications");

-				m_ThreadInfo->send(iq);

-				return true;

-			}

-			else {

-				// if it's something else than 'session-initiate' and noone processed it yet, reply with "unknown-session"

-				XmlNodeIq iq("error", idStr, from);

-				TiXmlElement *errNode = iq << XCHILD("error");

-				errNode << XATTR("type", "cancel");

-				errNode << XCHILDNS("item-not-found", "urn:ietf:params:xml:ns:xmpp-stanzas");

-				errNode << XCHILDNS("unknown-session", "urn:xmpp:jingle:errors:1");

-				m_ThreadInfo->send(iq);

-				return true;

+			const char *szInitiator = XmlGetAttr(child, "initiator");

+			auto *content = XmlGetChildByTag(child, "content", "creator", "initiator");

+

+			if (szAction && szSid) {

+				if (!mir_strcmp(szAction, "session-initiate")) {

+					// if this is a Jingle 'session-initiate' and noone processed it yet, reply with "unsupported-applications"

+					m_ThreadInfo->send(XmlNodeIq("result", idStr, from));

+

+					const TiXmlElement *descr = XmlGetChildByTag(content, "description", "xmlns", JABBER_FEAT_JINGLE_RTP);

+					if (m_bEnableVOIP && m_voipSession == "" && descr) {

+						if (VOIPCallAccept(child, from)) {

+							m_voipSession = szSid;

+							m_voipPeerJid = from;

+							return true;

+						}

+					}

+

+					XmlNodeIq iq("set", SerialNext(), from);

+					TiXmlElement *jingleNode = iq << XCHILDNS("jingle", JABBER_FEAT_JINGLE);

+					jingleNode << XATTR("action", "session-terminate");

+					if (szInitiator)

+						jingleNode << XATTR("initiator", szInitiator);

+					if (szSid)

+						jingleNode << XATTR("sid", szSid);

+					jingleNode << XCHILD("reason") << XCHILD("unsupported-applications");

+

+					m_ThreadInfo->send(iq);

+					return true;

+				}

+				else if (!mir_strcmp(szAction, "session-accept")) {

+					if (m_bEnableVOIP && m_voipSession == szSid) {

+						m_ThreadInfo->send(XmlNodeIq("result", idStr, from));

+						OnRTPDescription(child);

+						return true;

+					}

+				}

+				else if (!mir_strcmp(szAction, "session-terminate")) {

+					if (m_bEnableVOIP && m_voipSession == szSid) {

+						// EndCall()

+						m_ThreadInfo->send(XmlNodeIq("result", idStr, from));

+						VOIPTerminateSession();

+						m_voipSession.Empty();

+						m_voipPeerJid.Empty();

+						return true;

+					}

+				}

+				else if (!mir_strcmp(szAction, "transport-info")) {

+					auto *transport = XmlGetChildByTag(content, "transport", "xmlns", JABBER_FEAT_JINGLE_ICEUDP);

+					if (m_bEnableVOIP && m_voipSession == szSid && transport) {

+						m_ThreadInfo->send(XmlNodeIq("result", idStr, from));

+						if (const TiXmlElement *candidate = XmlFirstChild(transport, "candidate")) {

+							OnICECandidate(candidate, from);

+							return true;

+						}

+					}

+				}

 			}

+

+			// if it's something else than 'session-initiate' and noone processed it yet, reply with "unknown-session"

+			XmlNodeIq iq("error", idStr, from);

+			TiXmlElement *errNode = iq << XCHILD("error");

+			errNode << XATTR("type", "cancel");

+			errNode << XCHILDNS("item-not-found", "urn:ietf:params:xml:ns:xmpp-stanzas");

+			errNode << XCHILDNS("unknown-session", "urn:xmpp:jingle:errors:1");

+			m_ThreadInfo->send(iq);

+			return true;

 		}

 	}

 	return false;

diff --git a/protocols/JabberG/src/jabber_voip.cpp b/protocols/JabberG/src/jabber_voip.cpp
new file mode 100644
index 0000000000..2f45d61350
--- /dev/null
+++ b/protocols/JabberG/src/jabber_voip.cpp
@@ -0,0 +1,519 @@
+#include "stdafx.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",*/ NULL };
+
+	GstRegistry *registry = gst_registry_get();
+	gst_registry_scan_path(registry, "libs\\gst_plugins");
+	gboolean ret = TRUE;
+	for (int i = 0; needed[i]; i++) {
+		GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
+		if (!plugin) {
+			gst_print("Required gstreamer plugin '%s' not found\n", needed[i]);
+			ret = FALSE;
+		}
+		else gst_object_unref(plugin);
+	}
+
+	return ret;
+}
+
+void dbgprint(const gchar *string)
+{
+	OutputDebugStringA(string);
+}
+
+bool CJabberProto::VOIPCreatePipeline(void)
+{
+	//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);
+			return false;
+		}
+		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);
+			return false;
+		}
+		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)
+		MessageBoxA(0, "Epic fail", "cannot create m_webrtc1", MB_OK);
+
+	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)
+		goto err;
+
+	return true;
+
+err:
+	VOIPTerminateSession();
+	return FALSE;
+}
+
+bool CJabberProto::VOIPTerminateSession()
+{
+	gst_print("Terminating session");
+	if (m_pipe1) {
+		gst_element_set_state(GST_ELEMENT(m_pipe1), GST_STATE_NULL);
+		g_clear_object(&m_pipe1);
+		gst_print("Pipeline stopped\n");
+		gst_object_unref(m_pipe1);
+	}
+
+	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)
+{
+	// 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, const char *)
+{
+	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 != "") {
+		VOIPTerminateSession();
+		MessageBoxA(0, "Terminated", NULL, 0);
+		return 0;
+	}
+
+	CMStringA jid(ptrA(getUStringA(hContact, "jid")));
+	if (jid == "")
+		return 0;
+	ptrA szResource(GetBestResourceName(jid));
+	if (szResource)
+		jid = MakeJid(jid, szResource);
+
+	CMStringA question(FORMAT, "Call %s?\r\n"
+		"It will disclose IP address to the peer and his server", jid.c_str());
+	if (MessageBoxA(0, question.c_str(), "Outgoing call", MB_YESNO | MB_ICONQUESTION) != IDYES)
+		return 0;
+
+	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();
+
+	VOIPCreatePipeline();
+	return 0;
+}
+
+bool CJabberProto::VOIPCallAccept(const TiXmlElement *jingleNode, const char *from)
+{
+	if (!from || !jingleNode)
+		return false;
+
+	CMStringW question(FORMAT, TranslateT("Accept call from %s?\r\nIt will disclose IP address to the peer and his server"), from);
+	if (MessageBoxW(0, question, TranslateT("Incomig call"), MB_YESNO | MB_ICONQUESTION) != IDYES)
+		return false;
+
+	m_isOutgoing = false;
+
+	if (!VOIPCreatePipeline())
+		return false;
+
+	OnRTPDescription(jingleNode);
+
+	// ringing message
+	XmlNodeIq iq("set", SerialNext(), from);
+	TiXmlElement *rjNode = iq << XCHILDNS("jingle", JABBER_FEAT_JINGLE);
+	rjNode << XATTR("action", "session-info");
+	const char *szInitiator = XmlGetAttr(jingleNode, "initiator");
+	if (szInitiator)
+		rjNode << XATTR("initiator", szInitiator);
+	const char *szSid = XmlGetAttr(jingleNode, "sid");
+	if (szSid)
+		rjNode << XATTR("sid", szSid);
+	rjNode << XCHILDNS("ringing", "urn:xmpp:jingle:apps:rtp:info:1");
+
+	m_ThreadInfo->send(iq);
+	return true;
+}