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webrtc: Don't crash in ICE gathering

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Fix a crash collating ICE gathering states if there are
unassociated transceivers in the list with no TransportStream
This commit is contained in:
Jan Schmidt 2020-03-05 04:18:03 +11:00 committed by GStreamer Merge Bot
parent 469d2cac2f
commit ad53de1da1
1 changed files with 24 additions and 6 deletions
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30
ext/webrtc/gstwebrtcbin.c
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@ -905,12 +905,14 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
i);
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
TransportStream *stream = trans->stream;
GstWebRTCDTLSTransport *dtls_transport;
GstWebRTCICETransport *transport, *rtcp_transport;
GstWebRTCICEGatheringState ice_state;
gboolean rtcp_mux = FALSE;
if (rtp_trans->stopped) {
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
if (rtp_trans->stopped || stream == NULL) {
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated",
rtp_trans);
continue;
}
@ -922,7 +924,13 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
if (dtls_transport == NULL) {
GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans);
continue;
}
transport = dtls_transport->transport;
/* get gathering state */
g_object_get (transport, "gathering-state", &ice_state, NULL);
@ -932,8 +940,12 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
if (ice_state != STATE (COMPLETE))
all_completed = FALSE;
rtcp_transport =
webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans)->transport;
dtls_transport = webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans);
if (dtls_transport == NULL) {
GST_WARNING ("Transceiver %p has no DTLS RTCP transport", rtp_trans);
continue;
}
rtcp_transport = dtls_transport->transport;
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
g_object_get (rtcp_transport, "gathering-state", &ice_state, NULL);
@ -4751,6 +4763,9 @@ gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
NULL);
trans = _create_webrtc_transceiver (webrtc, direction, -1);
GST_LOG_OBJECT (webrtc,
"Created new unassociated transceiver %" GST_PTR_FORMAT, trans);
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
if (caps)
rtp_trans->codec_preferences = gst_caps_ref (caps);
@ -5573,10 +5588,13 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, serial);
trans = _find_transceiver_for_mline (webrtc, serial);
if (!trans)
if (!trans) {
trans =
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, serial));
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT
" for mline %u", trans, serial);
}
pad->trans = gst_object_ref (trans);
pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK |