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webrtcbin: fix regression with missing RTP header extensions in Answer SDP

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webrtcsrc first creates recvonly transceivers with codec-preferences
and expects that after applying a remote description, the
previously created transceivers are used rather than having new
transceivers created.

When pairing webrtcsink + webrtcsrc, the offer sdp from webrtcsink has a media
section with sendonly direction. In !7156, which was implemented following
RFC9429 Section 5.10, we only reuse a unassociated transceiver when applying a
remote description if the media is sendrecv or recvonly, and that caused creation
of new transceivers when applying a remote offer in webrtcsrc, thus losing
information from codec preferences like the RTP extension headers in the
previously created transceivers.

Since the change in !7156 broke existing code from webrtcsrc, relax the condition
for reusing unassociated transceivers and add a test to document this behavior which
wasn't covered by any tests before.

Fixes #3753.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7417>
This commit is contained in:
Carlos Bentzen 2024-08-27 11:52:08 +02:00 committed by GStreamer Marge Bot
parent ac868d9dc1
commit 77faf0a163
2 changed files with 125 additions and 41 deletions
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79
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -6415,49 +6415,46 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
}
} else {
if (!trans) {
/* RFC9429: If the "m=" section is "sendrecv" or "recvonly", and there are RtpTransceivers of the same type
* that were added to the PeerConnection by addTrack and are not associated with any "m=" section
* and are not stopped, find the first (according to the canonical order described in Section 5.2.1)
* such RtpTransceiver. */
if (direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV
|| direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY) {
int j;
for (j = 0; j < webrtc->priv->transceivers->len; ++j) {
trans = g_ptr_array_index (webrtc->priv->transceivers, j);
if (trans->mid || trans->stopped) {
trans = NULL;
continue;
}
/* FIXME: Here we shouldn't in theory need to match caps, as the spec says only about
* "RtpTransceivers of the same type". However, transceivers created by requesting sink
* pads (aka addTrack) may still have unknown type at this point. We may be missing updating
* the transceiver type early enough during caps negotation.
*/
GstCaps *trans_caps =
_find_codec_preferences (webrtc, trans, i, error);
if (error && *error)
goto out;
if (trans_caps) {
GstCaps *offer_caps = _rtp_caps_from_media (media);
GstCaps *caps = gst_caps_intersect (offer_caps, trans_caps);
gst_caps_unref (offer_caps);
gst_caps_unref (trans_caps);
if (caps) {
if (!gst_caps_is_empty (caps)) {
GST_LOG_OBJECT (webrtc,
"found compatible transceiver %" GST_PTR_FORMAT
" for offer media %u", trans, i);
gst_caps_unref (caps);
break;
}
gst_caps_unref (caps);
caps = NULL;
}
}
int j;
/* XXX: According to RFC9429 Section 5.10. we should only be finding compatible unassociated transceivers here if the
* media direction is "sendrecv" or "recvonly", but webrtcsrc and possibly other applications rely on this working
* also for "sendonly".
*/
for (j = 0; j < webrtc->priv->transceivers->len; ++j) {
trans = g_ptr_array_index (webrtc->priv->transceivers, j);
if (trans->mid || trans->stopped) {
trans = NULL;
continue;
}
/* FIXME: Here we shouldn't in theory need to match caps, as the spec says only about
* "RtpTransceivers of the same type". However, transceivers created by requesting sink
* pads (aka addTrack) may still have unknown type at this point. We may be missing updating
* the transceiver type early enough during caps negotation.
*/
GstCaps *trans_caps =
_find_codec_preferences (webrtc, trans, i, error);
if (error && *error)
goto out;
if (trans_caps) {
GstCaps *offer_caps = _rtp_caps_from_media (media);
GstCaps *caps = gst_caps_intersect (offer_caps, trans_caps);
gst_caps_unref (offer_caps);
gst_caps_unref (trans_caps);
if (caps) {
if (!gst_caps_is_empty (caps)) {
GST_LOG_OBJECT (webrtc,
"found compatible transceiver %" GST_PTR_FORMAT
" for offer media %u", trans, i);
gst_caps_unref (caps);
break;
}
gst_caps_unref (caps);
caps = NULL;
}
}
trans = NULL;
}
}

87
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
View file

@ -4906,6 +4906,92 @@ GST_START_TEST (test_audio_sendrecv)
GST_END_TEST;
static void
on_sdp_media_rtp_header_extensions (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GArray *expected_extensions = user_data;
int i;
for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0
|| g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
int extension_idx = 0;
int j;
for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
if (g_strcmp0 (attr->key, "extmap") == 0) {
GStrv split = g_strsplit (attr->value, " ", 2);
fail_unless_equals_string (split[1],
g_array_index (expected_extensions, char *, extension_idx++));
g_strfreev (split);
}
}
fail_unless_equals_int (expected_extensions->len, extension_idx);
}
}
}
#define TWCC_URI "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
GST_START_TEST (test_rtp_header_extension_sendonly_recvonly_pair)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
GstWebRTCRTPTransceiver *trans;
GstCaps *caps;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
caps =
gst_caps_from_string (OPUS_RTP_CAPS (96) ", extmap-1=(string)" TWCC_URI);
GstStructure *s = gst_caps_get_structure (caps, 0);
gst_structure_set (s, "ssrc", G_TYPE_UINT, 0xDEADBEEF, NULL);
gst_structure_set (s, "payload", G_TYPE_INT, 96, NULL);
gst_harness_set_src_caps (h, gst_caps_copy (caps));
gst_harness_add_src_parse (h, "fakesrc is-live=true", TRUE);
t->harnesses = g_list_prepend (t->harnesses, h);
g_signal_emit_by_name (t->webrtc1, "get-transceiver", 0, &trans);
g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
gst_object_unref (trans);
g_signal_emit_by_name (t->webrtc2, "add-transceiver",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, caps, &trans);
fail_unless (trans != NULL);
gst_object_unref (trans);
gst_caps_unref (caps);
const gchar *expected_offer_direction[] = { "sendonly", };
VAL_SDP_INIT (offer_direction, on_sdp_media_direction,
expected_offer_direction, NULL);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer_direction, on_sdp_media_direction,
expected_answer_direction, NULL);
const gchar *EXPECTED_EXTENSIONS_DATA[] = { TWCC_URI, };
GArray *expected_extensions = g_array_new (FALSE, FALSE, sizeof (gchar *));
g_array_append_vals (expected_extensions, EXPECTED_EXTENSIONS_DATA,
sizeof (EXPECTED_EXTENSIONS_DATA) / sizeof (gchar *));
VAL_SDP_INIT (offer, on_sdp_media_rtp_header_extensions, expected_extensions,
&offer_direction);
VAL_SDP_INIT (answer, on_sdp_media_rtp_header_extensions, expected_extensions,
&answer_direction);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
g_array_free (expected_extensions, TRUE);
}
GST_END_TEST;
static void
new_jitterbuffer_set_fast_start (GstElement * rtpbin,
GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
@ -6440,6 +6526,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_msid);
tcase_add_test (tc, test_ice_end_of_candidates);
tcase_add_test (tc, test_sdp_session_setup_attribute);
tcase_add_test (tc, test_rtp_header_extension_sendonly_recvonly_pair);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_create_two_channels);