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webrtcbin: Validate locked m-lines in set*Description

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Verify that the remote description match the locked m-lines, otherwise
just reject the SDP.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-03-26 20:02:13 -04:00
parent be84cc2c54
commit 2ca4cea538
1 changed files with 58 additions and 0 deletions
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58
ext/webrtc/gstwebrtcbin.c
View file

@ -4716,6 +4716,61 @@ check_transceivers_not_removed (GstWebRTCBin * webrtc,
return TRUE;
}
static gboolean
check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp,
GError ** error)
{
guint i;
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
GstWebRTCRTPTransceiver *rtp_trans;
WebRTCTransceiver *trans;
rtp_trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i);
/* only look for matching mid */
if (rtp_trans == NULL)
continue;
trans = WEBRTC_TRANSCEIVER (rtp_trans);
/* We only validate the locked mlines for now */
if (!trans->mline_locked)
continue;
if (rtp_trans->mline != i) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
"m-line with mid %s is at position %d, but was locked to %d, "
"rejecting", rtp_trans->mid, i, rtp_trans->mline);
return FALSE;
}
if (rtp_trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio") &&
rtp_trans->kind != GST_WEBRTC_KIND_AUDIO) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
"m-line %d was locked to audio, but SDP has %s media", i,
gst_sdp_media_get_media (media));
return FALSE;
}
if (!g_strcmp0 (gst_sdp_media_get_media (media), "video") &&
rtp_trans->kind != GST_WEBRTC_KIND_VIDEO) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
"m-line %d was locked to video, but SDP has %s media", i,
gst_sdp_media_get_media (media));
return FALSE;
}
}
}
return TRUE;
}
struct set_description
{
GstPromise *promise;
@ -4797,6 +4852,9 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
goto out;
}
if (!check_locked_mlines (webrtc, sd->sdp, &error))
goto out;
switch (sd->sdp->type) {
case GST_WEBRTC_SDP_TYPE_OFFER:{
if (sd->source == SDP_LOCAL) {