diff options
Diffstat (limited to 'plugins/Jingle/src/voip.cpp')
| -rw-r--r-- | plugins/Jingle/src/voip.cpp | 525 |
1 files changed, 525 insertions, 0 deletions
diff --git a/plugins/Jingle/src/voip.cpp b/plugins/Jingle/src/voip.cpp new file mode 100644 index 0000000000..bd5e541499 --- /dev/null +++ b/plugins/Jingle/src/voip.cpp @@ -0,0 +1,525 @@ +#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> + +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; + CJabberAccount *jproto = (CJabberAccount *)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->m_api->GetSerialNext(), 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_api->GetFullJid() : jproto->m_voipPeerJid); + if (!outgoing) + rjNode << XATTR("responder", jproto->m_api->GetFullJid()); + + 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_api->SendXml(iq); + + gst_webrtc_session_description_free(offer); +} + +void on_negotiation_needed_cb(GstElement *webrtcbin, gpointer user_data) +{ + if (((CJabberAccount *)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(((CJabberAccount *)user_data)->m_webrtc1, "create-answer", NULL, promise); +} + +void send_ice_candidate_message_cb(G_GNUC_UNUSED GstElement */*webrtcbin*/, guint mline_index, gchar *candidate, CJabberAccount *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->m_api->GetSerialNext(), 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_api->SendXml(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 CJabberAccount::VOIPCreatePipeline(void) +{ + if (!m_bEnableVOIP) + 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 CJabberAccount::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 (reason && !m_voipSession.IsEmpty() && !m_voipPeerJid.IsEmpty()) { + XmlNodeIq iq("set", m_api->GetSerialNext(), 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_api->GetFullJid() : m_voipPeerJid); + jingleNode << XCHILD("reason") << XCHILD(reason); + + m_api->SendXml(iq); + } + + m_voipICEPwd.Empty(); + m_voipICEUfrag.Empty(); + m_medianame.Empty(); + + m_voipSession.Empty(); + m_voipPeerJid.Empty(); + m_pipe1 = m_webrtc1 = NULL; + return true; +} + +bool CJabberAccount::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 CJabberAccount::OnICECandidate(const TiXmlElement *Node) +{ + if (!m_bEnableVOIP) + 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 CJabberAccount::VOIPCallIinitiate(MCONTACT hContact) +{ + if (!m_voipSession.IsEmpty()) { + VOIPTerminateSession(); + MessageBoxA(0, "Something went wrong\r\nOld session terminated", NULL, 0); + return false; + } + + if (!m_bEnableVOIP) + return false; + + CMStringA jid(db_get_sm(hContact, m_szModuleName, "jid")); + if (jid.IsEmpty()) + return false; + + ptrA szResource(m_api->GetBestResourceName(jid)); + if (szResource) { + jid.AppendFormat("/%s", szResource.get()); + bool bFound = false; + ptrA szFeatures(m_api->GetResourceFeatures(jid)); + for (auto *p = szFeatures.get(); *p; p += mir_strlen(p)) + if (!mir_strcmp(p, JABBER_FEAT_JINGLE)) + bFound = true; + + if (!bFound) { + // MsgPopup(hContact, TranslateT("Client's program does not support voice calls"), TranslateT("Error")); + return false; + } + } + + 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; +} + +///////////////////////////////////////////////////////////////////////////////////////// +// module entry point + +void CJabberAccount::InitVoip(bool bEnable) +{ + // Voip + VOICE_MODULE vsr = {}; + vsr.cbSize = sizeof(VOICE_MODULE); + vsr.description = L"XMPP/DTLS-SRTP"; + vsr.name = (char*)m_szModuleName; + vsr.icon = g_plugin.getIconHandle(IDI_MAIN); + vsr.flags = 3; + if (bEnable) + CallService(MS_VOICESERVICE_REGISTER, (WPARAM)&vsr, 0); + else { + VOIPTerminateSession(); + CallService(MS_VOICESERVICE_UNREGISTER, (WPARAM)&vsr, 0); + } +} |