/* * Demo gstreamer app for negotiating and streaming a sendrecv audio-only webrtc * stream to all the peers in a multiparty room. * * gcc mp-webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o mp-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 */ ROOM_JOINING = 3000, ROOM_JOIN_ERROR, ROOM_JOINED, ROOM_CALL_NEGOTIATING = 4000, /* negotiating with some or all peers */ ROOM_CALL_OFFERING, /* when we're the one sending the offer */ ROOM_CALL_ANSWERING, /* when we're the one answering an offer */ ROOM_CALL_STARTED, /* in a call with some or all peers */ ROOM_CALL_STOPPING, ROOM_CALL_STOPPED, ROOM_CALL_ERROR, }; static GMainLoop *loop; static GstElement *pipeline; static GList *peers; static SoupWebsocketConnection *ws_conn = NULL; static enum AppState app_state = 0; static const gchar *default_server_url = "wss://webrtc.nirbheek.in:8443"; static gchar *server_url = NULL; static gchar *local_id = NULL; static gchar *room_id = NULL; static gboolean strict_ssl = TRUE; static GOptionEntry entries[] = { {"name", 0, 0, G_OPTION_ARG_STRING, &local_id, "Name we will send to the server", "ID"}, {"room-id", 0, 0, G_OPTION_ARG_STRING, &room_id, "Room name to join or create", "ID"}, {"server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL"}, {NULL} }; static gint compare_str_glist (gconstpointer a, gconstpointer b) { return g_strcmp0 (a, b); } static const gchar * find_peer_from_list (const gchar * peer_id) { return (g_list_find_custom (peers, peer_id, compare_str_glist))->data; } static gboolean cleanup_and_quit_loop (const gchar * msg, enum AppState state) { if (msg) gst_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, *sink; GstPadLinkReturn ret; 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); 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_cmpint (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)) { gst_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 { gst_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad)); } } static void on_incoming_stream (GstElement * webrtc, GstPad * pad, GstElement * pipe) { GstElement *decodebin; GstPad *sinkpad; 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); sinkpad = gst_element_get_static_pad (decodebin, "sink"); gst_pad_link (pad, sinkpad); gst_object_unref (sinkpad); } static void send_room_peer_msg (const gchar * text, const gchar * peer_id) { gchar *msg; msg = g_strdup_printf ("ROOM_PEER_MSG %s %s", peer_id, text); soup_websocket_connection_send_text (ws_conn, msg); g_free (msg); } static void send_ice_candidate_message (GstElement * webrtc G_GNUC_UNUSED, guint mlineindex, gchar * candidate, const gchar * peer_id) { gchar *text; JsonObject *ice, *msg; if (app_state < ROOM_CALL_OFFERING) { 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); send_room_peer_msg (text, peer_id); g_free (text); } static void send_room_peer_sdp (GstWebRTCSessionDescription * desc, const gchar * peer_id) { JsonObject *msg, *sdp; gchar *text, *sdptype, *sdptext; g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING); if (desc->type == GST_WEBRTC_SDP_TYPE_OFFER) sdptype = "offer"; else if (desc->type == GST_WEBRTC_SDP_TYPE_ANSWER) sdptype = "answer"; else g_assert_not_reached (); text = gst_sdp_message_as_text (desc->sdp); gst_print ("Sending sdp %s to %s:\n%s\n", sdptype, peer_id, text); sdp = json_object_new (); json_object_set_string_member (sdp, "type", sdptype); json_object_set_string_member (sdp, "sdp", text); g_free (text); msg = json_object_new (); json_object_set_object_member (msg, "sdp", sdp); sdptext = get_string_from_json_object (msg); json_object_unref (msg); send_room_peer_msg (sdptext, peer_id); g_free (sdptext); } /* Offer created by our pipeline, to be sent to the peer */ static void on_offer_created (GstPromise * promise, const gchar * peer_id) { GstElement *webrtc; GstWebRTCSessionDescription *offer; const GstStructure *reply; g_assert_cmpint (app_state, ==, ROOM_CALL_OFFERING); g_assert_cmpint (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 (); webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id); g_assert_nonnull (webrtc); g_signal_emit_by_name (webrtc, "set-local-description", offer, promise); gst_promise_interrupt (promise); gst_promise_unref (promise); /* Send offer to peer */ send_room_peer_sdp (offer, peer_id); gst_webrtc_session_description_free (offer); } static void on_negotiation_needed (GstElement * webrtc, const gchar * peer_id) { GstPromise *promise; app_state = ROOM_CALL_OFFERING; promise = gst_promise_new_with_change_func ( (GstPromiseChangeFunc) on_offer_created, (gpointer) peer_id, NULL); g_signal_emit_by_name (webrtc, "create-offer", NULL, promise); } static void remove_peer_from_pipeline (const gchar * peer_id) { gchar *qname; GstPad *srcpad, *sinkpad; GstElement *webrtc, *q, *tee; webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id); if (!webrtc) return; gst_bin_remove (GST_BIN (pipeline), webrtc); gst_object_unref (webrtc); qname = g_strdup_printf ("queue-%s", peer_id); q = gst_bin_get_by_name (GST_BIN (pipeline), qname); g_free (qname); sinkpad = gst_element_get_static_pad (q, "sink"); g_assert_nonnull (sinkpad); srcpad = gst_pad_get_peer (sinkpad); g_assert_nonnull (srcpad); gst_object_unref (sinkpad); gst_bin_remove (GST_BIN (pipeline), q); gst_object_unref (q); tee = gst_bin_get_by_name (GST_BIN (pipeline), "audiotee"); g_assert_nonnull (tee); gst_element_release_request_pad (tee, srcpad); gst_object_unref (srcpad); gst_object_unref (tee); } static void add_peer_to_pipeline (const gchar * peer_id, gboolean offer) { int ret; gchar *tmp; GstElement *tee, *webrtc, *q; GstPad *srcpad, *sinkpad; tmp = g_strdup_printf ("queue-%s", peer_id); q = gst_element_factory_make ("queue", tmp); g_free (tmp); webrtc = gst_element_factory_make ("webrtcbin", peer_id); gst_bin_add_many (GST_BIN (pipeline), q, webrtc, NULL); srcpad = gst_element_get_static_pad (q, "src"); g_assert_nonnull (srcpad); sinkpad = gst_element_request_pad_simple (webrtc, "sink_%u"); g_assert_nonnull (sinkpad); ret = gst_pad_link (srcpad, sinkpad); g_assert_cmpint (ret, ==, GST_PAD_LINK_OK); gst_object_unref (srcpad); gst_object_unref (sinkpad); tee = gst_bin_get_by_name (GST_BIN (pipeline), "audiotee"); g_assert_nonnull (tee); srcpad = gst_element_request_pad_simple (tee, "src_%u"); g_assert_nonnull (srcpad); gst_object_unref (tee); sinkpad = gst_element_get_static_pad (q, "sink"); g_assert_nonnull (sinkpad); ret = gst_pad_link (srcpad, sinkpad); g_assert_cmpint (ret, ==, GST_PAD_LINK_OK); gst_object_unref (srcpad); gst_object_unref (sinkpad); /* This is the gstwebrtc entry point where we create the offer and so on. It * will be called when the pipeline goes to PLAYING. * XXX: We must connect this after webrtcbin has been linked to a source via * get_request_pad() and before we go from NULL->READY otherwise webrtcbin * will create an SDP offer with no media lines in it. */ if (offer) g_signal_connect (webrtc, "on-negotiation-needed", G_CALLBACK (on_negotiation_needed), (gpointer) peer_id); /* 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 (webrtc, "on-ice-candidate", G_CALLBACK (send_ice_candidate_message), (gpointer) peer_id); /* Incoming streams will be exposed via this signal */ g_signal_connect (webrtc, "pad-added", G_CALLBACK (on_incoming_stream), pipeline); /* Set to pipeline branch to PLAYING */ ret = gst_element_sync_state_with_parent (q); g_assert_true (ret); ret = gst_element_sync_state_with_parent (webrtc); g_assert_true (ret); } static void call_peer (const gchar * peer_id) { add_peer_to_pipeline (peer_id, TRUE); } static void incoming_call_from_peer (const gchar * peer_id) { add_peer_to_pipeline (peer_id, FALSE); } #define STR(x) #x #define RTP_CAPS_OPUS(x) "application/x-rtp,media=audio,encoding-name=OPUS,payload=" STR(x) static gboolean start_pipeline (void) { GstStateChangeReturn ret; GError *error = NULL; /* NOTE: webrtcbin currently does not support dynamic addition/removal of * streams, so we use a separate webrtcbin for each peer, but all of them are * inside the same pipeline. We start by connecting it to a fakesink so that * we can preroll early. */ pipeline = gst_parse_launch ("tee name=audiotee ! queue ! fakesink " "audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay ! " "queue ! " RTP_CAPS_OPUS (96) " ! audiotee. ", &error); if (error) { gst_printerr ("Failed to parse launch: %s\n", error->message); g_error_free (error); goto err; } gst_print ("Starting pipeline, not transmitting yet\n"); ret = gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING); if (ret == GST_STATE_CHANGE_FAILURE) goto err; return TRUE; err: gst_print ("State change failure\n"); if (pipeline) g_clear_object (&pipeline); return FALSE; } static gboolean join_room_on_server (void) { gchar *msg; if (soup_websocket_connection_get_state (ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) return FALSE; if (!room_id) return FALSE; gst_print ("Joining room %s\n", room_id); app_state = ROOM_JOINING; msg = g_strdup_printf ("ROOM %s", room_id); soup_websocket_connection_send_text (ws_conn, msg); g_free (msg); return TRUE; } static gboolean register_with_server (void) { gchar *hello; if (soup_websocket_connection_get_state (ws_conn) != SOUP_WEBSOCKET_STATE_OPEN) return FALSE; gst_print ("Registering id %s with server\n", local_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 %s", local_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); } static gboolean do_registration (void) { if (app_state != SERVER_REGISTERING) { cleanup_and_quit_loop ("ERROR: Received HELLO when not registering", APP_STATE_ERROR); return FALSE; } app_state = SERVER_REGISTERED; gst_print ("Registered with server\n"); /* Ask signalling server that we want to join a room */ if (!join_room_on_server ()) { cleanup_and_quit_loop ("ERROR: Failed to join room", ROOM_CALL_ERROR); return FALSE; } return TRUE; } /* * When we join a room, we are responsible for calling by starting negotiation * with each peer in it by sending an SDP offer and ICE candidates. */ static void do_join_room (const gchar * text) { gint ii, len; gchar **peer_ids; if (app_state != ROOM_JOINING) { cleanup_and_quit_loop ("ERROR: Received ROOM_OK when not calling", ROOM_JOIN_ERROR); return; } app_state = ROOM_JOINED; gst_print ("Room joined\n"); /* Start recording, but not transmitting */ if (!start_pipeline ()) { cleanup_and_quit_loop ("ERROR: Failed to start pipeline", ROOM_CALL_ERROR); return; } peer_ids = g_strsplit (text, " ", -1); g_assert_cmpstr (peer_ids[0], ==, "ROOM_OK"); len = g_strv_length (peer_ids); /* There are peers in the room already. We need to start negotiation * (exchange SDP and ICE candidates) and transmission of media. */ if (len > 1 && strlen (peer_ids[1]) > 0) { gst_print ("Found %i peers already in room\n", len - 1); app_state = ROOM_CALL_OFFERING; for (ii = 1; ii < len; ii++) { gchar *peer_id = g_strdup (peer_ids[ii]); gst_print ("Negotiating with peer %s\n", peer_id); /* This might fail asynchronously */ call_peer (peer_id); peers = g_list_prepend (peers, peer_id); } } g_strfreev (peer_ids); return; } static void handle_error_message (const gchar * msg) { switch (app_state) { case SERVER_CONNECTING: app_state = SERVER_CONNECTION_ERROR; break; case SERVER_REGISTERING: app_state = SERVER_REGISTRATION_ERROR; break; case ROOM_JOINING: app_state = ROOM_JOIN_ERROR; break; case ROOM_JOINED: case ROOM_CALL_NEGOTIATING: case ROOM_CALL_OFFERING: case ROOM_CALL_ANSWERING: app_state = ROOM_CALL_ERROR; break; case ROOM_CALL_STARTED: case ROOM_CALL_STOPPING: case ROOM_CALL_STOPPED: app_state = ROOM_CALL_ERROR; break; default: app_state = APP_STATE_ERROR; } cleanup_and_quit_loop (msg, 0); } static void on_answer_created (GstPromise * promise, const gchar * peer_id) { GstElement *webrtc; GstWebRTCSessionDescription *answer; const GstStructure *reply; g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING); g_assert_cmpint (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED); reply = gst_promise_get_reply (promise); gst_structure_get (reply, "answer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL); gst_promise_unref (promise); promise = gst_promise_new (); webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id); g_assert_nonnull (webrtc); g_signal_emit_by_name (webrtc, "set-local-description", answer, promise); gst_promise_interrupt (promise); gst_promise_unref (promise); /* Send offer to peer */ send_room_peer_sdp (answer, peer_id); gst_webrtc_session_description_free (answer); app_state = ROOM_CALL_STARTED; } static void handle_sdp_offer (const gchar * peer_id, const gchar * text) { int ret; GstPromise *promise; GstElement *webrtc; GstSDPMessage *sdp; GstWebRTCSessionDescription *offer; g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING); gst_print ("Received offer:\n%s\n", text); ret = gst_sdp_message_new (&sdp); g_assert_cmpint (ret, ==, GST_SDP_OK); ret = gst_sdp_message_parse_buffer ((guint8 *) text, strlen (text), sdp); g_assert_cmpint (ret, ==, GST_SDP_OK); offer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, sdp); g_assert_nonnull (offer); /* Set remote description on our pipeline */ promise = gst_promise_new (); webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id); g_assert_nonnull (webrtc); g_signal_emit_by_name (webrtc, "set-remote-description", offer, promise); /* We don't want to be notified when the action is done */ gst_promise_interrupt (promise); gst_promise_unref (promise); /* Create an answer that we will send back to the peer */ promise = gst_promise_new_with_change_func ( (GstPromiseChangeFunc) on_answer_created, (gpointer) peer_id, NULL); g_signal_emit_by_name (webrtc, "create-answer", NULL, promise); gst_webrtc_session_description_free (offer); gst_object_unref (webrtc); } static void handle_sdp_answer (const gchar * peer_id, const gchar * text) { int ret; GstPromise *promise; GstElement *webrtc; GstSDPMessage *sdp; GstWebRTCSessionDescription *answer; g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING); gst_print ("Received answer:\n%s\n", text); ret = gst_sdp_message_new (&sdp); g_assert_cmpint (ret, ==, GST_SDP_OK); ret = gst_sdp_message_parse_buffer ((guint8 *) text, strlen (text), sdp); g_assert_cmpint (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 */ promise = gst_promise_new (); webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id); g_assert_nonnull (webrtc); g_signal_emit_by_name (webrtc, "set-remote-description", answer, promise); gst_object_unref (webrtc); /* We don't want to be notified when the action is done */ gst_promise_interrupt (promise); gst_promise_unref (promise); } static gboolean handle_peer_message (const gchar * peer_id, const gchar * msg) { JsonNode *root; JsonObject *object, *child; JsonParser *parser = json_parser_new (); if (!json_parser_load_from_data (parser, msg, -1, NULL)) { gst_printerr ("Unknown message '%s' from '%s', ignoring", msg, peer_id); g_object_unref (parser); return FALSE; } root = json_parser_get_root (parser); if (!JSON_NODE_HOLDS_OBJECT (root)) { gst_printerr ("Unknown json message '%s' from '%s', ignoring", msg, peer_id); g_object_unref (parser); return FALSE; } gst_print ("Message from peer %s: %s\n", peer_id, msg); object = json_node_get_object (root); /* Check type of JSON message */ if (json_object_has_member (object, "sdp")) { const gchar *text, *sdp_type; g_assert_cmpint (app_state, >=, ROOM_JOINED); child = json_object_get_object_member (object, "sdp"); if (!json_object_has_member (child, "type")) { cleanup_and_quit_loop ("ERROR: received SDP without 'type'", ROOM_CALL_ERROR); return FALSE; } sdp_type = json_object_get_string_member (child, "type"); text = json_object_get_string_member (child, "sdp"); if (g_strcmp0 (sdp_type, "offer") == 0) { app_state = ROOM_CALL_ANSWERING; incoming_call_from_peer (peer_id); handle_sdp_offer (peer_id, text); } else if (g_strcmp0 (sdp_type, "answer") == 0) { g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING); handle_sdp_answer (peer_id, text); app_state = ROOM_CALL_STARTED; } else { cleanup_and_quit_loop ("ERROR: invalid sdp_type", ROOM_CALL_ERROR); return FALSE; } } else if (json_object_has_member (object, "ice")) { GstElement *webrtc; 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 */ webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id); g_assert_nonnull (webrtc); g_signal_emit_by_name (webrtc, "add-ice-candidate", sdpmlineindex, candidate); gst_object_unref (webrtc); } else { gst_printerr ("Ignoring unknown JSON message:\n%s\n", msg); } g_object_unref (parser); return TRUE; } /* One mega message handler for our asynchronous calling mechanism */ static void on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, GBytes * message, gpointer user_data) { gchar *text; switch (type) { case SOUP_WEBSOCKET_DATA_BINARY: gst_printerr ("Received unknown binary message, ignoring\n"); return; case SOUP_WEBSOCKET_DATA_TEXT:{ gsize size; const gchar *data = g_bytes_get_data (message, &size); /* Convert to NULL-terminated string */ text = g_strndup (data, size); break; } default: g_assert_not_reached (); } /* Server has accepted our registration, we are ready to send commands */ if (g_strcmp0 (text, "HELLO") == 0) { /* May fail asynchronously */ do_registration (); /* Room-related message */ } else if (g_str_has_prefix (text, "ROOM_")) { /* Room joined, now we can start negotiation */ if (g_str_has_prefix (text, "ROOM_OK ")) { /* May fail asynchronously */ do_join_room (text); } else if (g_str_has_prefix (text, "ROOM_PEER")) { gchar **splitm = NULL; const gchar *peer_id; /* SDP and ICE, usually */ if (g_str_has_prefix (text, "ROOM_PEER_MSG")) { splitm = g_strsplit (text, " ", 3); peer_id = find_peer_from_list (splitm[1]); g_assert_nonnull (peer_id); /* Could be an offer or an answer, or ICE, or an arbitrary message */ handle_peer_message (peer_id, splitm[2]); } else if (g_str_has_prefix (text, "ROOM_PEER_JOINED")) { splitm = g_strsplit (text, " ", 2); peers = g_list_prepend (peers, g_strdup (splitm[1])); peer_id = find_peer_from_list (splitm[1]); g_assert_nonnull (peer_id); gst_print ("Peer %s has joined the room\n", peer_id); } else if (g_str_has_prefix (text, "ROOM_PEER_LEFT")) { splitm = g_strsplit (text, " ", 2); peer_id = find_peer_from_list (splitm[1]); g_assert_nonnull (peer_id); peers = g_list_remove (peers, peer_id); gst_print ("Peer %s has left the room\n", peer_id); remove_peer_from_pipeline (peer_id); g_free ((gchar *) peer_id); /* TODO: cleanup pipeline */ } else { gst_printerr ("WARNING: Ignoring unknown message %s\n", text); } g_strfreev (splitm); } else { goto err; } /* Handle errors */ } else if (g_str_has_prefix (text, "ERROR")) { handle_error_message (text); } else { goto err; } out: g_free (text); return; err: { gchar *err_s = g_strdup_printf ("ERROR: unknown message %s", text); cleanup_and_quit_loop (err_s, 0); g_free (err_s); goto out; } } 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; gst_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 * responses from the server will be handled in on_server_message() above */ 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, strict_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); gst_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; GstRegistry *registry; const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp", "rtpmanager", "audiotestsrc", NULL }; registry = gst_registry_get (); ret = TRUE; for (i = 0; i < g_strv_length ((gchar **) needed); i++) { GstPlugin *plugin; plugin = gst_registry_find_plugin (registry, needed[i]); if (!plugin) { gst_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)) { gst_printerr ("Error initializing: %s\n", error->message); return -1; } if (!check_plugins ()) return -1; if (!room_id) { gst_printerr ("--room-id is a required argument\n"); return -1; } if (!local_id) local_id = g_strdup_printf ("%s-%i", g_get_user_name (), g_random_int_range (10, 10000)); /* Sanitize by removing whitespace, modifies string in-place */ g_strdelimit (local_id, " \t\n\r", '-'); gst_print ("Our local id is %s\n", local_id); if (!server_url) server_url = g_strdup (default_server_url); /* Don't use strict 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) strict_ssl = FALSE; gst_uri_unref (uri); } loop = g_main_loop_new (NULL, FALSE); connect_to_websocket_server_async (); g_main_loop_run (loop); gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL); gst_print ("Pipeline stopped\n"); gst_object_unref (pipeline); g_free (server_url); g_free (local_id); g_free (room_id); return 0; }