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Handle ICE failure

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This commit is contained in:
trilene 2020-07-26 10:59:50 -04:00
parent 57d5a3d31f
commit 43ec0c0624
7 changed files with 131 additions and 73 deletions
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65
src/WebRTCSession.cpp
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@ -14,9 +14,9 @@ extern "C" {
Q_DECLARE_METATYPE(WebRTCSession::State)
namespace {
bool gisoffer;
std::string glocalsdp;
std::vector<mtx::events::msg::CallCandidates::Candidate> gcandidates;
bool isoffering_;
std::string localsdp_;
std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data);
GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type);
@ -24,6 +24,7 @@ void generateOffer(GstElement *webrtc);
void setLocalDescription(GstPromise *promise, gpointer webrtc);
void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED);
gboolean onICEGatheringCompletion(gpointer timerid);
void iceConnectionStateChanged(GstElement *webrtcbin, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED);
void createAnswer(GstPromise *promise, gpointer webrtc);
void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
@ -92,9 +93,9 @@ WebRTCSession::init(std::string *errorMessage)
bool
WebRTCSession::createOffer()
{
gisoffer = true;
glocalsdp.clear();
gcandidates.clear();
isoffering_ = true;
localsdp_.clear();
localcandidates_.clear();
return startPipeline(111); // a dynamic opus payload type
}
@ -105,9 +106,9 @@ WebRTCSession::acceptOffer(const std::string &sdp)
if (state_ != State::DISCONNECTED)
return false;
gisoffer = false;
glocalsdp.clear();
gcandidates.clear();
isoffering_ = false;
localsdp_.clear();
localcandidates_.clear();
int opusPayloadType = getPayloadType(sdp, "opus");
if (opusPayloadType == -1)
@ -152,14 +153,20 @@ WebRTCSession::startPipeline(int opusPayloadType)
gboolean udata;
g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
}
if (turnServers_.empty())
nhlog::ui()->warn("WebRTC: no TURN server provided");
// generate the offer when the pipeline goes to PLAYING
if (gisoffer)
if (isoffering_)
g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr);
// on-ice-candidate is emitted when a local ICE candidate has been gathered
g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
// capture ICE failure
g_signal_connect(webrtc_, "notify::ice-connection-state",
G_CALLBACK(iceConnectionStateChanged), nullptr);
// incoming streams trigger pad-added
gst_element_set_state(pipe_, GST_STATE_READY);
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
@ -229,8 +236,6 @@ WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandi
nhlog::ui()->debug("WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
}
if (state_ == State::OFFERSENT)
emit stateChanged(State::CONNECTING);
}
}
@ -357,11 +362,11 @@ setLocalDescription(GstPromise *promise, gpointer webrtc)
g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
glocalsdp = std::string(sdp);
localsdp_ = std::string(sdp);
g_free(sdp);
gst_webrtc_session_description_free(gstsdp);
nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", glocalsdp);
nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
}
void
@ -369,12 +374,12 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *
{
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
if (WebRTCSession::instance().state() == WebRTCSession::State::CONNECTED) {
if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
return;
}
gcandidates.push_back({"audio", (uint16_t)mlineIndex, candidate});
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
// fixed in v1.18
@ -390,18 +395,36 @@ gboolean
onICEGatheringCompletion(gpointer timerid)
{
*(guint*)(timerid) = 0;
if (gisoffer) {
emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
if (isoffering_) {
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
}
else {
emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
}
return FALSE;
}
void
iceConnectionStateChanged(GstElement *webrtc, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED)
{
GstWebRTCICEConnectionState newState;
g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
switch (newState) {
case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
break;
case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
break;
default:
break;
}
}
void
createAnswer(GstPromise *promise, gpointer webrtc)
{