diff options
| -rw-r--r-- | src/WebRTCSession.cpp | 8 |
1 files changed, 5 insertions, 3 deletions
diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp index 2248fb1a..b4e7eeb3 100644 --- a/src/WebRTCSession.cpp +++ b/src/WebRTCSession.cpp @@ -223,18 +223,19 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, { nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate); +#if GST_CHECK_VERSION(1, 17, 0) + localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate}); + return; +#else if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) { emit WebRTCSession::instance().newICECandidate( {"audio", (uint16_t)mlineIndex, candidate}); return; } - 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 v1.17. // Use a 100ms timeout in the meantime -#if !GST_CHECK_VERSION(1, 17, 0) static guint timerid = 0; if (timerid) g_source_remove(timerid); @@ -447,6 +448,7 @@ WebRTCSession::startPipeline(int opusPayloadType) g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr); } + for (const auto &uri : turnServers_) { nhlog::ui()->info("WebRTC: setting TURN server: {}", uri); gboolean udata; |