/* * Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream * with a browser JS app. * * gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv * * Author: Nirbheek Chauhan */ #include #include #define GST_USE_UNSTABLE_API #include /* For signalling */ #include #include #include enum AppState { APP_STATE_UNKNOWN = 0, APP_STATE_ERROR = 1, /* generic error */ SERVER_CONNECTING = 1000, SERVER_CONNECTION_ERROR, SERVER_CONNECTED, /* Ready to register */ SERVER_REGISTERING = 2000, SERVER_REGISTRATION_ERROR, SERVER_REGISTERED, /* Ready to call a peer */ SERVER_CLOSED, /* server connection closed by us or the server */ PEER_CONNECTING = 3000, PEER_CONNECTION_ERROR, PEER_CONNECTED, PEER_CALL_NEGOTIATING = 4000, PEER_CALL_STARTED, PEER_CALL_STOPPING, PEER_CALL_STOPPED, PEER_CALL_ERROR, }; static GMainLoop *loop; static GstElement *pipe1, *webrtc1; static SoupWebsocketConnection *ws_conn = NULL; static enum AppState app_state = 0; static const gchar *peer_id = NULL; static const gchar *server_url = "wss://webrtc.nirbheek.in:8443"; static gboolean disable_ssl = FALSE; static GOptionEntry entries[] = { { "peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id, "String ID of the peer to connect to", "ID" }, { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" }, { "disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL }, { NULL }, }; static gboolean cleanup_and_quit_loop (const gchar * msg, enum AppState state) { if (msg) g_printerr ("%s\n", msg); if (state > 0) app_state = state; if (ws_conn) { if (soup_websocket_connection_get_state (ws_conn) == SOUP_WEBSOCKET_STATE_OPEN) /* This will call us again */ soup_websocket_connection_close (ws_conn, 1000, ""); else g_object_unref (ws_conn); } if (loop) { g_main_loop_quit (loop); loop = NULL; } /* To allow usage as a GSourceFunc */ return G_SOURCE_REMOVE; } static gchar* get_string_from_json_object (JsonObject * object) { JsonNode *root; JsonGenerator *generator; gchar *text; /* Make it the root node */ root = json_node_init_object (json_node_alloc (), object); generator = json_generator_new (); json_generator_set_root (generator, root); text = json_generator_to_data (generator, NULL); /* Release everything */ g_object_unref (generator); json_node_free (root); return text; } static void handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name, const char * sink_name) { GstPad *qpad; GstElement *q, *conv, *resample, *sink; GstPadLinkReturn ret; g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name); q = gst_element_factory_make ("queue", NULL); g_assert_nonnull (q); conv = gst_element_factory_make (convert_name, NULL); g_assert_nonnull (conv); sink = gst_element_factory_make (sink_name, NULL); g_assert_nonnull (sink); if (g_strcmp0 (convert_name, "audioconvert") == 0) { /* Might also need to resample, so add it just in case. * Will be a no-op if it's not required. */ resample = gst_element_factory_make ("audioresample", NULL); g_assert_nonnull (resample); gst_bin_add_many (GST_BIN (pipe), q, conv, resample, sink, NULL); gst_element_sync_state_with_parent (q); gst_element_sync_state_with_parent (conv); gst_element_sync_state_with_parent (resample); gst_element_sync_state_with_parent (sink); gst_element_link_many (q, conv, resample, sink, NULL); } else { gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL); gst_element_sync_state_with_parent (q); gst_element_sync_state_with_parent (conv); gst_element_sync_state_with_parent (sink); gst_element_link_many (q, conv, sink, NULL); } qpad = gst_element_get_static_pad (q, "sink"); ret = gst_pad_link (pad, qpad); g_assert_cmphex (ret, ==, GST_PAD_LINK_OK); } static void on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad, GstElement * pipe) { GstCaps *caps; const gchar *name; if (!gst_pad_has_current_caps (pad)) { g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n", GST_PAD_NAME (pad)); return; } caps = gst_pad_get_current_caps (pad); name = gst_structure_get_name (gst_caps_get_structure (caps, 0)); if (g_str_has_prefix (name, "video")) { handle_media_stream (pad, pipe, "videoconvert", "autovideosink"); } else if (g_str_has_prefix (name, "audio")) { handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink"); } else { g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad)); } } static void on_incoming_stream (GstElement * webrtc, GstPad * pad, GstElement * pipe) { GstElement *decodebin; if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC) return; decodebin = gst_element_factory_make ("decodebin", NULL); g_signal_connect (decodebin, "pad-added", G_CALLBACK (on_incoming_decodebin_stream), pipe); gst_bin_add (GST_BIN (pipe), decodebin); gst_element_sync_state_with_parent (decodebin); gst_element_link (webrtc, decodebin); } static void send_ice_candidate_message (GstElement * webrtc G_GNUC_UNUSED, guint mlineindex, gchar * candidate, gpointer user_data G_GNUC_UNUSED) { gchar *text; JsonObject *ice, *msg; if (app_state < PEER_CALL_NEGOTIATING) { cleanup_and_quit_loop ("Can't send ICE, not in call", APP_STATE_ERROR); return; } ice = json_object_new (); json_object_set_string_member (ice, "candidate", candidate); json_object_set_int_member (ice, "sdpMLineIndex", mlineindex); msg = json_object_new (); json_object_set_object_member (msg, "ice", ice); text = get_string_from_json_object (msg); json_object_unref (msg); soup_websocket_connection_send_text (ws_conn, text); g_free (text); } static void send_sdp_offer (GstWebRTCSessionDescription * offer) { gchar *text; JsonObject *msg, *sdp; if (app_state < PEER_CALL_NEGOTIATING) { cleanup_and_quit_loop ("Can't send offer, not in call", APP_STATE_ERROR); return; } text = gst_sdp_message_as_text (offer->sdp); g_print ("Sending offer:\n%s\n", text); sdp = json_object_new (); json_object_set_string_member (sdp, "type", "offer"); json_object_set_string_member (sdp, "sdp", text); g_free (text); msg = json_object_new (); json_object_set_object_member (msg, "sdp", sdp); text = get_string_from_json_object (msg); json_object_unref (msg); soup_websocket_connection_send_text (ws_conn, text); g_free (text); } /* Offer created by our pipeline, to be sent to the peer */ static void on_offer_created (GstPromise * promise, gpointer user_data) { GstWebRTCSessionDescription *offer = NULL; const GstStructure *reply; g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING); g_assert_cmphex (gst_promise_wait(promise), ==, GST_PROMISE_RESULT_REPLIED); reply = gst_promise_get_reply (promise); gst_structure_get (reply, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL); gst_promise_unref (promise); promise = gst_promise_new (); g_signal_emit_by_name (webrtc1, "set-local-description", offer, promise); gst_promise_interrupt (promise); gst_promise_unref (promise); /* Send offer to peer */ send_sdp_offer (offer); gst_webrtc_session_description_free (offer); } static void on_negotiation_needed (GstElement * element, gpointer user_data) { GstPromise *promise; app_state = PEER_CALL_NEGOTIATING; promise = gst_promise_new_with_change_func (on_offer_created, user_data, NULL);; g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise); } #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 gboolean start_pipeline (void) { GstStateChangeReturn ret; GError *error = NULL; pipe1 = gst_parse_launch ("webrtcbin name=sendrecv " STUN_SERVER "videotestsrc pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! " "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. " "audiotestsrc wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! " "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error); if (error) { g_printerr ("Failed to parse launch: %s\n", error->message); g_error_free (error); goto err; } webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv"); g_assert_nonnull (webrtc1); /* This is the gstwebrtc entry point where we create the offer and so on. It * will be called when the pipeline goes to PLAYING. */ g_signal_connect (webrtc1, "on-negotiation-needed", G_CALLBACK (on_negotiation_needed), NULL); /* We need to transmit this ICE candidate to the browser via the websockets * signalling server. Incoming ice candidates from the browser need to be * added by us too, see on_server_message() */ g_signal_connect (webrtc1, "on-ice-candidate", G_CALLBACK (send_ice_candidate_message), NULL); /* Incoming streams will be exposed via this signal */ g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream), pipe1); /* Lifetime is the same as the pipeline itself */ gst_object_unref (webrtc1); g_print ("Starting pipeline\n"); ret = gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING); if (ret == GST_STATE_CHANGE_FAILURE) goto err; return TRUE; err: if (pipe1) g_clear_object (&pipe1); if (webrtc1) webrtc1 = NULL; return FALSE; } static gboolean setup_call (void) { gchar *msg; if (soup_websocket_connection_get_state (ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) return FALSE; if (!peer_id) return FALSE; g_print ("Setting up signalling server call with %s\n", peer_id); app_state = PEER_CONNECTING; msg = g_strdup_printf ("SESSION %s", peer_id); soup_websocket_connection_send_text (ws_conn, msg); g_free (msg); return TRUE; } static gboolean register_with_server (void) { gchar *hello; gint32 our_id; if (soup_websocket_connection_get_state (ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) return FALSE; our_id = g_random_int_range (10, 10000); g_print ("Registering id %i with server\n", our_id); app_state = SERVER_REGISTERING; /* Register with the server with a random integer id. Reply will be received * by on_server_message() */ hello = g_strdup_printf ("HELLO %i", our_id); soup_websocket_connection_send_text (ws_conn, hello); g_free (hello); return TRUE; } static void on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED) { app_state = SERVER_CLOSED; cleanup_and_quit_loop ("Server connection closed", 0); } /* One mega message handler for our asynchronous calling mechanism */ static void on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, GBytes * message, gpointer user_data) { gsize size; gchar *text, *data; switch (type) { case SOUP_WEBSOCKET_DATA_BINARY: g_printerr ("Received unknown binary message, ignoring\n"); g_bytes_unref (message); return; case SOUP_WEBSOCKET_DATA_TEXT: data = g_bytes_unref_to_data (message, &size); /* Convert to NULL-terminated string */ text = g_strndup (data, size); g_free (data); break; default: g_assert_not_reached (); } /* Server has accepted our registration, we are ready to send commands */ if (g_strcmp0 (text, "HELLO") == 0) { if (app_state != SERVER_REGISTERING) { cleanup_and_quit_loop ("ERROR: Received HELLO when not registering", APP_STATE_ERROR); goto out; } app_state = SERVER_REGISTERED; g_print ("Registered with server\n"); /* Ask signalling server to connect us with a specific peer */ if (!setup_call ()) { cleanup_and_quit_loop ("ERROR: Failed to setup call", PEER_CALL_ERROR); goto out; } /* Call has been setup by the server, now we can start negotiation */ } else if (g_strcmp0 (text, "SESSION_OK") == 0) { if (app_state != PEER_CONNECTING) { cleanup_and_quit_loop ("ERROR: Received SESSION_OK when not calling", PEER_CONNECTION_ERROR); goto out; } app_state = PEER_CONNECTED; /* Start negotiation (exchange SDP and ICE candidates) */ if (!start_pipeline ()) cleanup_and_quit_loop ("ERROR: failed to start pipeline", PEER_CALL_ERROR); /* Handle errors */ } else if (g_str_has_prefix (text, "ERROR")) { switch (app_state) { case SERVER_CONNECTING: app_state = SERVER_CONNECTION_ERROR; break; case SERVER_REGISTERING: app_state = SERVER_REGISTRATION_ERROR; break; case PEER_CONNECTING: app_state = PEER_CONNECTION_ERROR; break; case PEER_CONNECTED: case PEER_CALL_NEGOTIATING: app_state = PEER_CALL_ERROR; default: app_state = APP_STATE_ERROR; } cleanup_and_quit_loop (text, 0); /* Look for JSON messages containing SDP and ICE candidates */ } else { JsonNode *root; JsonObject *object, *child; JsonParser *parser = json_parser_new (); if (!json_parser_load_from_data (parser, text, -1, NULL)) { g_printerr ("Unknown message '%s', ignoring", text); g_object_unref (parser); goto out; } root = json_parser_get_root (parser); if (!JSON_NODE_HOLDS_OBJECT (root)) { g_printerr ("Unknown json message '%s', ignoring", text); g_object_unref (parser); goto out; } object = json_node_get_object (root); /* Check type of JSON message */ if (json_object_has_member (object, "sdp")) { int ret; GstSDPMessage *sdp; const gchar *text, *sdptype; GstWebRTCSessionDescription *answer; g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING); child = json_object_get_object_member (object, "sdp"); if (!json_object_has_member (child, "type")) { cleanup_and_quit_loop ("ERROR: received SDP without 'type'", PEER_CALL_ERROR); goto out; } sdptype = json_object_get_string_member (child, "type"); /* In this example, we always create the offer and receive one answer. * See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to * handle offers from peers and reply with answers using webrtcbin. */ g_assert_cmpstr (sdptype, ==, "answer"); text = json_object_get_string_member (child, "sdp"); g_print ("Received answer:\n%s\n", text); ret = gst_sdp_message_new (&sdp); g_assert_cmphex (ret, ==, GST_SDP_OK); ret = gst_sdp_message_parse_buffer ((guint8 *) text, strlen (text), sdp); g_assert_cmphex (ret, ==, GST_SDP_OK); answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, sdp); g_assert_nonnull (answer); /* Set remote description on our pipeline */ { GstPromise *promise = gst_promise_new (); g_signal_emit_by_name (webrtc1, "set-remote-description", answer, promise); gst_promise_interrupt (promise); gst_promise_unref (promise); } app_state = PEER_CALL_STARTED; } else if (json_object_has_member (object, "ice")) { const gchar *candidate; gint sdpmlineindex; child = json_object_get_object_member (object, "ice"); candidate = json_object_get_string_member (child, "candidate"); sdpmlineindex = json_object_get_int_member (child, "sdpMLineIndex"); /* Add ice candidate sent by remote peer */ g_signal_emit_by_name (webrtc1, "add-ice-candidate", sdpmlineindex, candidate); } else { g_printerr ("Ignoring unknown JSON message:\n%s\n", text); } g_object_unref (parser); } out: g_free (text); } static void on_server_connected (SoupSession * session, GAsyncResult * res, SoupMessage *msg) { GError *error = NULL; ws_conn = soup_session_websocket_connect_finish (session, res, &error); if (error) { cleanup_and_quit_loop (error->message, SERVER_CONNECTION_ERROR); g_error_free (error); return; } g_assert_nonnull (ws_conn); app_state = SERVER_CONNECTED; g_print ("Connected to signalling server\n"); g_signal_connect (ws_conn, "closed", G_CALLBACK (on_server_closed), NULL); g_signal_connect (ws_conn, "message", G_CALLBACK (on_server_message), NULL); /* Register with the server so it knows about us and can accept commands */ register_with_server (); } /* * Connect to the signalling server. This is the entrypoint for everything else. */ static void connect_to_websocket_server_async (void) { SoupLogger *logger; SoupMessage *message; SoupSession *session; const char *https_aliases[] = {"wss", NULL}; session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, !disable_ssl, SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE, //SOUP_SESSION_SSL_CA_FILE, "/etc/ssl/certs/ca-bundle.crt", SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL); logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1); soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger)); g_object_unref (logger); message = soup_message_new (SOUP_METHOD_GET, server_url); g_print ("Connecting to server...\n"); /* Once connected, we will register */ soup_session_websocket_connect_async (session, message, NULL, NULL, NULL, (GAsyncReadyCallback) on_server_connected, message); app_state = SERVER_CONNECTING; } static gboolean check_plugins (void) { int i; gboolean ret; GstPlugin *plugin; GstRegistry *registry; const gchar *needed[] = { "opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtpmanager", "videotestsrc", "audiotestsrc", NULL}; registry = gst_registry_get (); ret = TRUE; for (i = 0; i < g_strv_length ((gchar **) needed); i++) { plugin = gst_registry_find_plugin (registry, needed[i]); if (!plugin) { g_print ("Required gstreamer plugin '%s' not found\n", needed[i]); ret = FALSE; continue; } gst_object_unref (plugin); } return ret; } int main (int argc, char *argv[]) { GOptionContext *context; GError *error = NULL; context = g_option_context_new ("- gstreamer webrtc sendrecv demo"); g_option_context_add_main_entries (context, entries, NULL); g_option_context_add_group (context, gst_init_get_option_group ()); if (!g_option_context_parse (context, &argc, &argv, &error)) { g_printerr ("Error initializing: %s\n", error->message); return -1; } if (!check_plugins ()) return -1; if (!peer_id) { g_printerr ("--peer-id is a required argument\n"); return -1; } /* Disable ssl when running a localhost server, because * it's probably a test server with a self-signed certificate */ { GstUri *uri = gst_uri_from_string (server_url); if (g_strcmp0 ("localhost", gst_uri_get_host (uri)) == 0 || g_strcmp0 ("127.0.0.1", gst_uri_get_host (uri)) == 0) disable_ssl = TRUE; gst_uri_unref (uri); } loop = g_main_loop_new (NULL, FALSE); connect_to_websocket_server_async (); g_main_loop_run (loop); g_main_loop_unref (loop); if (pipe1) { gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL); g_print ("Pipeline stopped\n"); gst_object_unref (pipe1); } return 0; }