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webrtcbin: Make mid optional in offers and answers

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The mid attribute is not strictly required. Two new tests cover this change,
they remove the mid and group attributes from the SDP offers and answers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8729>
This commit is contained in:
Philippe Normand 2025-03-28 12:19:20 +00:00
parent 36c983ccd5
commit 18f6fec49f
3 changed files with 197 additions and 30 deletions
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27
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View File

@ -4540,8 +4540,6 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
}
mid = gst_sdp_media_get_attribute_val (media, "mid");
/* XXX: not strictly required but a lot of functionality requires a mid */
g_assert (mid);
/* set the a=setup: attribute */
offer_setup = _get_dtls_setup_from_media (offer_media);
@ -4603,6 +4601,7 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
_remove_optional_offer_fields (offer_caps);
if (mid) {
rtp_trans = _find_transceiver_for_mid (webrtc, mid);
if (!rtp_trans) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
@ -4610,6 +4609,15 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
gst_caps_unref (offer_caps);
goto rejected;
}
} else {
rtp_trans = _find_transceiver_for_mline (webrtc, i);
if (!rtp_trans) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Transceiver for media with mline %u not found", i);
gst_caps_unref (offer_caps);
goto rejected;
}
}
GstCaps *current_caps =
_find_codec_preferences (webrtc, rtp_trans, i, error);
if (*error) {
@ -4623,6 +4631,7 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
const gchar *last_mid =
gst_sdp_media_get_attribute_val (last_media, "mid");
/* FIXME: assumes no shenanigans with recycling transceivers */
if (mid != last_mid)
g_assert (g_strcmp0 (mid, last_mid) == 0);
if (!current_caps)
current_caps = _rtp_caps_from_media (last_media);
@ -6433,11 +6442,10 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
mid = gst_sdp_media_get_attribute_val (media, "mid");
direction = _get_direction_from_media (media);
/* XXX: not strictly required but a lot of functionality requires a mid */
if (!mid) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Missing mid attribute in media");
goto out;
if (mid) {
trans = _find_transceiver_for_mid (webrtc, mid);
} else {
trans = _find_transceiver_for_mline (webrtc, i);
}
if (bundled)
@ -6445,7 +6453,6 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
else
transport_idx = i;
trans = _find_transceiver_for_mid (webrtc, mid);
if (sd->source == SDP_LOCAL) {
/* If the media description was not yet associated with an RTCRtpTransceiver object then run the following steps: */
@ -6469,8 +6476,10 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
}
trans->mline = i;
/* Set transceiver.[[Mid]] to transceiver.[[JsepMid]] */
g_free (trans->mid);
g_clear_pointer (&trans->mid, g_free);
if (mid) {
trans->mid = g_strdup (mid);
}
g_object_notify (G_OBJECT (trans), "mid");
/* If transceiver.[[Stopped]] is true, abort these sub steps */
if (trans->stopped)

20
subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
View File

@ -193,16 +193,10 @@ _media_has_attribute_key (const GstSDPMedia * media, const gchar * key)
}
static gboolean
_media_has_mid (const GstSDPMedia * media, guint media_idx, GError ** error)
_media_has_mid (const GstSDPMedia * media, guint media_idx)
{
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
if (IS_EMPTY_SDP_ATTRIBUTE (mid)) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'mid\' attribute",
media_idx);
return FALSE;
}
return TRUE;
return !IS_EMPTY_SDP_ATTRIBUTE (mid);
}
const gchar *
@ -308,11 +302,13 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
const gchar *mid;
gboolean media_in_bundle = FALSE;
if (!_media_has_mid (media, i, error))
goto fail;
if (_media_has_mid (media, i)) {
mid = gst_sdp_media_get_attribute_val (media, "mid");
media_in_bundle = is_bundle
&& g_strv_contains ((const gchar **) group_members, mid);
media_in_bundle =
is_bundle && g_strv_contains ((const gchar **) group_members, mid);
}
if (!_media_get_ice_ufrag (sdp->sdp, i)) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"media %u is missing or contains an empty \'ice-ufrag\' attribute",

162
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
View File

@ -5044,6 +5044,166 @@ a=group:BUNDLE \r\n\
GST_END_TEST;
static GstWebRTCSessionDescription *
remove_sdp_attributes (const GstWebRTCSessionDescription * desc, guint n_attrs,
const char *const *attributes)
{
GstSDPMessage *sdp;
guint total_medias = gst_sdp_message_medias_len (desc->sdp);
gst_sdp_message_copy (desc->sdp, &sdp);
for (guint i = 0; i < total_medias; i++) {
gst_sdp_message_remove_media (sdp, i);
}
for (guint i = 0; i < total_medias; i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
guint total_attributes = gst_sdp_media_attributes_len (media);
GstSDPMedia *new_media;
gst_sdp_media_copy (media, &new_media);
for (guint ii = 0; ii < total_attributes; ii++) {
const GstSDPAttribute *attribute =
gst_sdp_media_get_attribute (new_media, ii);
for (guint iii = 0; iii < n_attrs; iii++) {
if (!g_strcmp0 (attribute->key, attributes[iii])) {
gst_sdp_media_remove_attribute (new_media, ii);
ii--;
total_attributes--;
break;
}
}
}
gst_sdp_message_add_media (sdp, new_media);
gst_sdp_media_free (new_media);
}
return gst_webrtc_session_description_new (desc->type, sdp);
}
static void
do_missing_mid_test (gboolean is_offer)
{
struct test_webrtc *t = test_webrtc_new ();
const gchar *attributes_to_remove[] = { "mid", "group" };
GstWebRTCSessionDescription *modified_desc = NULL;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GstWebRTCSessionDescription *desc;
GstHarness *h1;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
h1 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_audio_test_src_harness (h1, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h1);
fail_if (gst_element_set_state (t->webrtc1, GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2, GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE);
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
fail_if (gst_structure_has_field (s, "error"));
gst_structure_get (s, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &desc,
NULL);
fail_unless (desc != NULL);
gst_promise_unref (promise);
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "set-local-description", desc, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_if (s && gst_structure_has_field (s, "error"));
gst_promise_unref (promise);
if (is_offer) {
modified_desc =
remove_sdp_attributes (desc, G_N_ELEMENTS (attributes_to_remove),
attributes_to_remove);
}
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc2, "set-remote-description",
modified_desc ? modified_desc : desc, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_if (s && gst_structure_has_field (s, "error"));
gst_promise_unref (promise);
g_clear_pointer (&modified_desc, gst_webrtc_session_description_free);
gst_webrtc_session_description_free (desc);
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc2, "create-answer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
fail_if (gst_structure_has_field (s, "error"));
gst_structure_get (s, "answer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &desc,
NULL);
fail_unless (desc != NULL);
gst_promise_unref (promise);
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc2, "set-local-description", desc, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_if (s && gst_structure_has_field (s, "error"));
gst_promise_unref (promise);
if (!is_offer) {
modified_desc =
remove_sdp_attributes (desc, G_N_ELEMENTS (attributes_to_remove),
attributes_to_remove);
}
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "set-remote-description",
modified_desc ? modified_desc : desc, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_if (s && gst_structure_has_field (s, "error"));
gst_promise_unref (promise);
g_clear_pointer (&modified_desc, gst_webrtc_session_description_free);
gst_webrtc_session_description_free (desc);
fail_if (gst_element_set_state (t->webrtc1, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE);
test_webrtc_free (t);
}
GST_START_TEST (test_missing_mid_in_offer)
{
do_missing_mid_test (TRUE);
}
GST_END_TEST;
GST_START_TEST (test_missing_mid_in_answer)
{
do_missing_mid_test (FALSE);
}
GST_END_TEST;
static void
new_jitterbuffer_set_fast_start (GstElement * rtpbin,
GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
@ -6624,6 +6784,8 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_sdp_session_setup_attribute);
tcase_add_test (tc, test_rtp_header_extension_sendonly_recvonly_pair);
tcase_add_test (tc, test_invalid_bundle_in_pending_remote_description);
tcase_add_test (tc, test_missing_mid_in_offer);
tcase_add_test (tc, test_missing_mid_in_answer);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_create_two_channels);