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webrtcbin: Prevent crash when attempting to set answer on invalid SDP
If the pending remote description has an invalid BUNDLE group _parse_bundle() triggers early return from _create_answer_task(), before ret has been initialized, so it needs to be checked before attempting to call gst_sdp_message_copy(). Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7423>
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2 changed files with 54 additions and 1 deletions
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@ -4769,7 +4769,7 @@ out:
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webrtc->priv->last_generated_offer = NULL;
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if (webrtc->priv->last_generated_answer)
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gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
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{
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if (ret) {
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GstSDPMessage *copy;
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gst_sdp_message_copy (ret, ©);
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webrtc->priv->last_generated_answer =
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@ -4992,6 +4992,58 @@ GST_START_TEST (test_rtp_header_extension_sendonly_recvonly_pair)
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GST_END_TEST;
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GST_START_TEST (test_invalid_bundle_in_pending_remote_description)
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{
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GstPromise *promise;
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struct test_webrtc *t = test_webrtc_new ();
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const gchar *invalid_bundle = "v=0\r\n\
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o=thisisadapterortc 2683876491 2 IN IP4 127.0.0.1\r\n\
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s=-\r\n\
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t=0 0\r\n\
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a=setup:actpass\r\n\
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a=fingerprint:sha-256 95:B3:DB:24:83:3B:9E:3F:B0:AD:93:2D:EF:73:C9:D2:1C:68:EA:19:C6:F8:73:BA:9A:FA:34:A9:64:69:C0:D8\r\n\
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a=ice-ufrag:ERn4TYI2HSbtKrzQNCdp9wD2EHt4wM2O\r\n\
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a=ice-pwd:QxEZFuCPRwIURJPMSYNNYFr2XFNgqNkG\r\n\
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a=group:BUNDLE \r\n\
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";
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GstSDPMessage *sdp;
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const GstStructure *reply;
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GError *error = NULL;
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t->on_negotiation_needed = NULL;
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t->on_offer_created = NULL;
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t->on_answer_created = NULL;
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gst_sdp_message_new_from_text (invalid_bundle, &sdp);
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GstWebRTCSessionDescription *desc =
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gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER,
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sdp);
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gst_element_set_state (t->webrtc1, GST_STATE_READY);
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promise = gst_promise_new ();
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g_signal_emit_by_name (t->webrtc1, "set-remote-description", desc, promise);
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gst_promise_wait (promise);
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gst_promise_unref (promise);
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gst_webrtc_session_description_free (desc);
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/* Creating an answer from SDP with invalid BUNDLE group should trigger no crash. */
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promise = gst_promise_new ();
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g_signal_emit_by_name (t->webrtc1, "create-answer", NULL, promise);
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gst_promise_wait (promise);
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reply = gst_promise_get_reply (promise);
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fail_unless (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL));
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fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
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GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR));
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fail_unless_equals_string (error->message,
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"Invalid format for BUNDLE group, expected at least one mid (BUNDLE )");
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g_clear_error (&error);
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gst_promise_unref (promise);
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test_webrtc_free (t);
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}
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GST_END_TEST;
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static void
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new_jitterbuffer_set_fast_start (GstElement * rtpbin,
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GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
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@ -6527,6 +6579,7 @@ webrtcbin_suite (void)
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tcase_add_test (tc, test_ice_end_of_candidates);
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tcase_add_test (tc, test_sdp_session_setup_attribute);
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tcase_add_test (tc, test_rtp_header_extension_sendonly_recvonly_pair);
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tcase_add_test (tc, test_invalid_bundle_in_pending_remote_description);
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if (sctpenc && sctpdec) {
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tcase_add_test (tc, test_data_channel_create);
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tcase_add_test (tc, test_data_channel_create_two_channels);
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