gstreamer webrtcbin TURN STURN only get typ host candidate?

[孙建希]

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`.