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webrtc: don't crash on invalid bundle id

Browse source 
If the bundle id forwarded to connect_rtpfunnel is not valid,
the assertion fails and crashes the program. This is now instead
an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7963>
This commit is contained in:
Albert Sjolund 2024-11-26 09:23:51 +01:00
parent 7008e76c62
commit c31cbda5de
1 changed files with 11 additions and 5 deletions
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16
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -6046,15 +6046,19 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
transport_receive_bin_set_receive_state (receive, RECEIVE_STATE_PASS);
}
static void
_connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
static gboolean
_connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id, GError ** error)
{
gchar *pad_name;
GstPad *srcpad;
GstPad *rtp_sink;
TransportStream *stream = _find_transport_for_session (webrtc, session_id);
g_assert (stream);
if (!stream) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Invalid bundle id %u, no session found", session_id);
return FALSE;
}
if (webrtc->rtpfunnel)
goto done;
@ -6084,7 +6088,7 @@ _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
g_free (pad_name);
done:
return;
return TRUE;
}
static gboolean
@ -6126,7 +6130,9 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
}
ensure_rtx_hdr_ext (bundle_stream);
_connect_rtpfunnel (webrtc, bundle_idx);
if (!_connect_rtpfunnel (webrtc, bundle_idx, error)) {
goto done;
}
}
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {