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webrtc-sendrecv: Fix create-answer caps negotiation

Browse source 
We need to parse the payload type map provided by the offer SDP and
set those values on the payloader, otherwise webrtcbin will create
a recvonly answer SDP and we won't send anything to the browser.

Fixed it for both C and Python sendrecv examples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1995>
This commit is contained in:
Nirbheek Chauhan 2022-03-15 16:31:56 +05:30 committed by GStreamer Marge Bot
parent c786551322
commit 3bc84aa7d5
2 changed files with 108 additions and 44 deletions
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87
subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
View file

@ -301,10 +301,6 @@ on_negotiation_needed (GstElement * element, gpointer user_data)
}
}
#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="
static void
data_channel_on_error (GObject * dc, gpointer user_data)
{
@ -421,16 +417,23 @@ webrtcbin_get_stats (GstElement * webrtcbin)
return G_SOURCE_REMOVE;
}
#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "
#define RTP_TWCC_URI "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS"
#define RTP_OPUS_DEFAULT_PT 97
#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8"
#define RTP_VP8_DEFAULT_PT 96
static gboolean
start_pipeline (gboolean create_offer)
start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
{
char *pipeline;
GstStateChangeReturn ret;
GError *error = NULL;
pipe1 =
gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv "
pipeline =
g_strdup_printf ("webrtcbin bundle-policy=max-bundle name=sendrecv "
STUN_SERVER
"videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! "
/* increase the default keyframe distance, browsers have really long
@ -441,10 +444,13 @@ start_pipeline (gboolean create_offer)
/* picture-id-mode=15-bit seems to make TWCC stats behave better, and
* fixes stuttery video playback in Chrome */
"rtpvp8pay name=videopay picture-id-mode=15-bit ! "
"queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
"queue ! %s,payload=%u ! sendrecv. "
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! "
"queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);
"queue ! %s,payload=%u ! sendrecv. ", RTP_CAPS_VP8, vp8_pt,
RTP_CAPS_OPUS, opus_pt);
pipe1 = gst_parse_launch (pipeline, &error);
g_free (pipeline);
if (error) {
gst_printerr ("Failed to parse launch: %s\n", error->message);
g_error_free (error);
@ -454,7 +460,7 @@ start_pipeline (gboolean create_offer)
webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
g_assert_nonnull (webrtc1);
if (remote_is_offerer) {
if (!create_offer) {
/* XXX: this will fail when the remote offers twcc as the extension id
* cannot currently be negotiated when receiving an offer.
*/
@ -630,6 +636,50 @@ on_offer_received (GstSDPMessage * sdp)
GstWebRTCSessionDescription *offer = NULL;
GstPromise *promise;
/* If we got an offer and we have no webrtcbin, we need to parse the SDP,
* get the payload types, then start the pipeline */
if (!webrtc1 && our_id) {
guint medias_len, formats_len;
guint opus_pt = 0, vp8_pt = 0;
gst_println ("Parsing offer to find payload types");
medias_len = gst_sdp_message_medias_len (sdp);
for (int i = 0; i < medias_len; i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, i);
formats_len = gst_sdp_media_formats_len (media);
for (int j = 0; j < formats_len; j++) {
guint pt;
GstCaps *caps;
GstStructure *s;
const char *fmt, *encoding_name;
fmt = gst_sdp_media_get_format (media, j);
if (g_strcmp0 (fmt, "webrtc-datachannel") == 0)
continue;
pt = atoi (fmt);
caps = gst_sdp_media_get_caps_from_media (media, pt);
s = gst_caps_get_structure (caps, 0);
encoding_name = gst_structure_get_string (s, "encoding-name");
if (vp8_pt == 0 && g_strcmp0 (encoding_name, "VP8") == 0)
vp8_pt = pt;
if (opus_pt == 0 && g_strcmp0 (encoding_name, "OPUS") == 0)
opus_pt = pt;
}
}
g_assert_cmpint (opus_pt, !=, 0);
g_assert_cmpint (vp8_pt, !=, 0);
gst_println ("Starting pipeline with opus pt: %u vp8 pt: %u", opus_pt,
vp8_pt);
if (!start_pipeline (FALSE, opus_pt, vp8_pt)) {
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
}
}
offer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, sdp);
g_assert_nonnull (offer);
@ -692,7 +742,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
app_state = PEER_CONNECTED;
/* Start negotiation (exchange SDP and ICE candidates) */
if (!start_pipeline (TRUE))
if (!start_pipeline (TRUE, RTP_OPUS_DEFAULT_PT, RTP_VP8_DEFAULT_PT))
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
} else if (g_strcmp0 (text, "OFFER_REQUEST") == 0) {
@ -702,7 +752,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
}
gst_print ("Received OFFER_REQUEST, sending offer\n");
/* Peer wants us to start negotiation (exchange SDP and ICE candidates) */
if (!start_pipeline (TRUE))
if (!start_pipeline (TRUE, RTP_OPUS_DEFAULT_PT, RTP_VP8_DEFAULT_PT))
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
} else if (g_str_has_prefix (text, "ERROR")) {
@ -743,17 +793,6 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
goto out;
}
/* If peer connection wasn't made yet and we are expecting peer will
* connect to us, launch pipeline at this moment */
if (!webrtc1 && our_id) {
if (!start_pipeline (FALSE)) {
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
}
app_state = PEER_CALL_NEGOTIATING;
}
object = json_node_get_object (root);
/* Check type of JSON message */
if (json_object_has_member (object, "sdp")) {
@ -762,7 +801,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
const gchar *text, *sdptype;
GstWebRTCSessionDescription *answer;
g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
app_state = PEER_CALL_NEGOTIATING;
child = json_object_get_object_member (object, "sdp");

65
subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
View file

@ -35,9 +35,9 @@ PIPELINE_DESC = '''
webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
queue ! application/x-rtp,media=video,encoding-name=VP8,payload={vp8_pt} ! sendrecv.
audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={opus_pt} ! sendrecv.
'''
from websockets.version import version as wsv
@ -51,6 +51,33 @@ def print_error(msg):
print(f'!!! {msg}', file=sys.stderr)
def get_payload_types(sdpmsg, video_encoding, audio_encoding):
'''
Find the payload types for the specified video and audio encoding.
Very simplistically finds the first payload type matching the encoding
name. More complex applications will want to match caps on
profile-level-id, packetization-mode, etc.
'''
video_pt = None
audio_pt = None
for i in range(0, sdpmsg.medias_len()):
media = sdpmsg.get_media(i)
for j in range(0, media.formats_len()):
fmt = media.get_format(j)
if fmt == 'webrtc-datachannel':
continue
pt = int(fmt)
caps = media.get_caps_from_media(pt)
s = caps.get_structure(0)
encoding_name = s['encoding-name']
if video_pt is None and encoding_name == video_encoding:
video_pt = pt
elif audio_pt is None and encoding_name == audio_encoding:
audio_pt = pt
return {video_encoding: video_pt, audio_encoding: audio_pt}
class WebRTCClient:
def __init__(self, loop, our_id, peer_id, server, remote_is_offerer):
self.conn = None
@ -114,10 +141,6 @@ class WebRTCClient:
print_status('Call was connected: creating offer')
promise = Gst.Promise.new_with_change_func(self.on_offer_created, None, None)
self.webrtc.emit('create-offer', None, promise)
elif self.remote_is_offerer:
# We are initiating the call, but we want the remote peer to create the offer
print_status('Call was connected: requesting remote peer for offer')
self.send_soon('OFFER_REQUEST')
def send_ice_candidate_message(self, _, mlineindex, candidate):
icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
@ -167,9 +190,9 @@ class WebRTCClient:
decodebin.sync_state_with_parent()
self.webrtc.link(decodebin)
def start_pipeline(self, create_offer=True):
def start_pipeline(self, create_offer=True, opus_pt=96, vp8_pt=97):
print_status(f'Creating pipeline, create_offer: {create_offer}')
self.pipe = Gst.parse_launch(PIPELINE_DESC)
self.pipe = Gst.parse_launch(PIPELINE_DESC.format(vp8_pt=vp8_pt, opus_pt=opus_pt))
self.webrtc = self.pipe.get_by_name('sendrecv')
self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed, create_offer)
self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
@ -192,7 +215,6 @@ class WebRTCClient:
self.webrtc.emit('create-answer', None, promise)
def handle_json(self, message):
assert (self.webrtc)
try:
msg = json.loads(message)
except json.decoder.JSONDecoderError:
@ -212,10 +234,21 @@ class WebRTCClient:
print_status('Received offer:\n%s' % sdp)
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
if not self.webrtc:
print_status('Incoming call: received an offer, creating pipeline')
pts = get_payload_types(sdpmsg, video_encoding='VP8', audio_encoding='OPUS')
assert('VP8' in pts)
assert('OPUS' in pts)
self.start_pipeline(create_offer=False, vp8_pt=pts['VP8'], opus_pt=pts['OPUS'])
assert(self.webrtc)
offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new_with_change_func(self.on_offer_set, None, None)
self.webrtc.emit('set-remote-description', offer, promise)
elif 'ice' in msg:
assert(self.webrtc)
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
@ -229,13 +262,6 @@ class WebRTCClient:
self.pipe = None
self.webrtc = None
def is_incoming_offer(self, msg):
if self.webrtc:
return False
if self.remote_is_offerer:
return True
return True
async def loop(self):
assert self.conn
async for message in self.conn:
@ -254,7 +280,9 @@ class WebRTCClient:
await self.setup_call()
elif message == 'SESSION_OK':
if self.remote_is_offerer:
self.start_pipeline(create_offer=False)
# We are initiating the call, but we want the remote peer to create the offer
print_status('Call was connected: requesting remote peer for offer')
await self.send('OFFER_REQUEST')
else:
self.start_pipeline()
elif message == 'OFFER_REQUEST':
@ -265,9 +293,6 @@ class WebRTCClient:
self.close_pipeline()
return 1
else:
if self.is_incoming_offer(message):
print_status('Incoming call: received an offer, creating pipeline')
self.start_pipeline(create_offer=False)
self.handle_json(message)
self.close_pipeline()
return 0