gstreamer webrtcbin TURN STURN only get typ host candidate? - webrtc

Question

Test gstreamer webrtcbin android example, in local network everything is ok.But cross network, webrtc paused. ice send from android is all typ host.
Got ice server: candidate:1 1 UDP 2015363327 127.0.0.1 42258 typ host index: 0
Got ice server: candidate:2 1 TCP 1015021823 127.0.0.1 9 typ host tcptype active index: 0
Got ice server: candidate:3 1 TCP 1010827519 127.0.0.1 36241 typ host tcptype passive index: 0
Got ice server: candidate:4 1 UDP 2015363583 10.0.2.16 40513 typ host index: 0
Got ice server: candidate:5 1 TCP 1015022079 10.0.2.16 9 typ host tcptype active index: 0
Got ice server: candidate:6 1 TCP 1010827775 10.0.2.16 52791 typ host tcptype passive index: 0
Got ice server: candidate:7 1 UDP 2015363839 10.0.2.15 38413 typ host index: 0
Got ice server: candidate:8 1 TCP 1015022335 10.0.2.15 9 typ host tcptype active index: 0
Got ice server: candidate:9 1 TCP 1010828031 10.0.2.15 42225 typ host tcptype passive index: 0
#define STUN_SERVER " stun-server=stun://47.104.15.123:3478 "
#define TURN_SERVER " turn-server=turn://jianxi:jianxi#47.104.15.123:3478 "
webrtc->pipe =
gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv "
STUN_SERVER TURN_SERVER
if set GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY, android will not send any ice candidate.
sturn turn server is okay. Can not catch stun packet with wireshark.
g_signal_connect (webrtc->webrtcbin, "on-ice-candidate",
G_CALLBACK (send_ice_candidate_message), webrtc);
g_signal_connect (webrtc->webrtcbin, "notify::ice-gathering-state",
G_CALLBACK (on_ice_gathering_state_notify), NULL);
g_signal_connect (webrtc->webrtcbin, "notify::ice-connection-state",
G_CALLBACK (on_ice_gathering_state_notify), NULL);
gstreamer get signal notify:
static void
on_ice_gathering_state_notify (GstElement * webrtcbin, GParamSpec * pspec,
gpointer user_data)
{
GstWebRTCICEConnectionState ice_connect_state;
GstWebRTCICEGatheringState ice_gather_state;
gchar *stunser = NULL, *turnser = NULL;
const gchar *new_state = "unknown";
g_object_get (webrtcbin, "stun-server", &stunser, NULL);
if(stunser) {
gst_print("stun-server: %s\n", stunser);
g_free(stunser);
}
g_object_get (webrtcbin, "turn-server", &turnser, NULL);
if(turnser) {
gst_print("turn-server: %s\n", turnser);
g_free(turnser);
}
g_object_get (webrtcbin, "ice-gathering-state", &ice_gather_state, NULL);
g_object_get (webrtcbin, "ice-connection-state", &ice_connect_state, NULL);
switch (ice_gather_state) {
case GST_WEBRTC_ICE_GATHERING_STATE_NEW:
new_state = "new";
break;
case GST_WEBRTC_ICE_GATHERING_STATE_GATHERING:
new_state = "gathering";
break;
case GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE:
new_state = "complete";
break;
}
gst_print ("ICE gathering state changed to %s, %d\n", new_state, ice_connect_state);
}
2022-11-23 11:35:50.239 1638-5461 GLib+stdout org.freedesktop.gstreamer.webrtc I stun-server: stun://47.104.15.123:3478
2022-11-23 11:35:50.239 1638-5461 GLib+stdout org.freedesktop.gstreamer.webrtc I turn-server: turn://jianxi:jianxi#47.104.15.123:3478
2022-11-23 11:35:50.239 1638-5461 GLib+stdout org.freedesktop.gstreamer.webrtc I ICE gathering state changed to complete, 4
ice end is 4 GST_WEBRTC_ICE_CONNECTION_STATE_FAILED.

Answer 1

In most cases, it means there are some issues with connecting to stun/turn server from your device. Either the host is unreachable via udp or there are issues with authentication.
The easiest way to test it is with trickle-ice page from your device's browser. Set address and creds of your stun/turn and check if there are some relay/srflx/prflx candidates.
If so, then it's a configuration issue on gstreamer-side.
If not, then try using tcpdump and inspect stun requests/responses. Maybe you're not getting the responses, or there are some error responses which might give you an idea of what's wrong.