Media over Websocket Channel Driver

* Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information. * res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs. * main/channel.c: Changed an obsolete nuisance error to a debug. * ARI channels: Updated externalMedia to include chan_websocket as a supported transport. UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket.
2025-09-04 03:50:31 +00:00 · 2025-04-28 10:39:50 -06:00
parent d5bd2b3ce9
commit 5963e624e2
8 changed files with 1672 additions and 54 deletions
--- a/channels/chan_websocket.c
+++ b/channels/chan_websocket.c
--- a/include/asterisk/http_websocket.h
+++ b/include/asterisk/http_websocket.h
@@ -77,6 +77,14 @@ enum ast_websocket_opcode {
 	AST_WEBSOCKET_OPCODE_CONTINUATION = 0x0, /*!< Continuation of a previous frame */
 };

+#ifdef LOW_MEMORY
+/*! \brief Size of the pre-determined buffer for WebSocket frames */
+#define AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE 8192
+#else
+/*! \brief Size of the pre-determined buffer for WebSocket frames */
+#define AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE 65535
+#endif
+
 /*!
 * \brief Opaque structure for WebSocket server.
 * \since 12
--- a/main/channel.c
+++ b/main/channel.c
@@ -3527,16 +3527,12 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio, int
 		 * The ast_waitfor() code records which of the channel's file
 		 * descriptors reported that data is available.  In theory,
 		 * ast_read() should only be called after ast_waitfor() reports
-		 * that a channel has data available for reading.  However,
-		 * there still may be some edge cases throughout the code where
-		 * ast_read() is called improperly.  This can potentially cause
-		 * problems, so if this is a developer build, make a lot of
-		 * noise if this happens so that it can be addressed.
-		 *
-		 * One of the potential problems is blocking on a dead channel.
+		 * that a channel has data available for reading but certain
+		 * situations with stasis and ARI could give a false indication.
+		 * For this reason, we don't stop any processing.
 		 */
 		if (ast_channel_fdno(chan) == -1) {
-			ast_log(LOG_ERROR,
+			ast_debug(3,
 				"ast_read() on chan '%s' called with no recorded file descriptor.\n",
 				ast_channel_name(chan));
 		}
--- a/res/ari/resource_channels.c
+++ b/res/ari/resource_channels.c
@@ -44,6 +44,7 @@
 #include "asterisk/dial.h"
 #include "asterisk/max_forwards.h"
 #include "asterisk/rtp_engine.h"
+#include "asterisk/websocket_client.h"
 #include "resource_channels.h"

 #include <limits.h>
@@ -2179,6 +2180,53 @@ static int external_media_audiosocket_tcp(struct ast_ari_channels_external_media
 	return 0;
 }

+static int external_media_websocket(struct ast_ari_channels_external_media_args *args,
+	struct ast_variable *variables,
+	struct ast_ari_response *response)
+{
+	char *endpoint;
+	struct ast_channel *chan;
+	struct varshead *vars;
+
+	if (ast_asprintf(&endpoint, "WebSocket/%s/c(%s)",
+			args->external_host,
+			args->format) == -1) {
+		return 1;
+	}
+
+	chan = ari_channels_handle_originate_with_id(
+		endpoint,
+		NULL,
+		NULL,
+		0,
+		NULL,
+		args->app,
+		args->data,
+		NULL,
+		0,
+		variables,
+		args->channel_id,
+		NULL,
+		NULL,
+		args->format,
+		response);
+
+	ast_free(endpoint);
+
+	if (!chan) {
+		return 1;
+	}
+
+	ast_channel_lock(chan);
+	vars = ast_channel_varshead(chan);
+	if (vars && !AST_LIST_EMPTY(vars)) {
+		ast_json_object_set(response->message, "channelvars", ast_json_channel_vars(vars));
+	}
+	ast_channel_unlock(chan);
+	ast_channel_unref(chan);
+	return 0;
+}
+
 #include "asterisk/config.h"
 #include "asterisk/netsock2.h"

@@ -2209,31 +2257,77 @@ void ast_ari_channels_external_media(struct ast_variable *headers,
 		return;
 	}

-	if (ast_strlen_zero(args->external_host)) {
-		ast_ari_response_error(response, 400, "Bad Request", "external_host cannot be empty");
-		return;
+	if (ast_strlen_zero(args->transport)) {
+		args->transport = "udp";
 	}

+	if (ast_strlen_zero(args->encapsulation)) {
+		args->encapsulation = "rtp";
+	}
+	if (ast_strings_equal(args->transport, "websocket")) {
+		if (!ast_strings_equal(args->encapsulation, "none")) {
+			ast_ari_response_error(response, 400, "Bad Request", "encapsulation must be 'none' for websocket transport");
+			return;
+		}
+	}
+
+	if (ast_strings_equal(args->encapsulation, "rtp")) {
+		if (!ast_strings_equal(args->transport, "udp")) {
+			ast_ari_response_error(response, 400, "Bad Request", "transport must be 'udp' for rtp encapsulation");
+			return;
+		}
+	}
+
+	if (ast_strings_equal(args->encapsulation, "audiosocket")) {
+		if (!ast_strings_equal(args->transport, "tcp")) {
+			ast_ari_response_error(response, 400, "Bad Request", "transport must be 'tcp' for audiosocket encapsulation");
+			return;
+		}
+	}
+
+	if (ast_strlen_zero(args->connection_type)) {
+		args->connection_type = "client";
+	}
+	if (!ast_strings_equal(args->transport, "websocket")) {
+		if (ast_strings_equal(args->connection_type, "server")) {
+			ast_ari_response_error(response, 400, "Bad Request", "'server' connection_type can only be used with the websocket transport");
+			return;
+		}
+	}
+
+	if (ast_strlen_zero(args->external_host)) {
+		if (ast_strings_equal(args->connection_type, "client")) {
+			ast_ari_response_error(response, 400, "Bad Request", "external_host is required for all but websocket server connections");
+			return;
+		} else {
+			/* server is only valid for websocket, enforced above */
+			args->external_host = "INCOMING";
+		}
+	}
+
+	if (ast_strings_equal(args->transport, "websocket")) {
+		if (ast_strings_equal(args->connection_type, "client")) {
+			struct ast_websocket_client *ws_client =
+				ast_websocket_client_retrieve_by_id(args->external_host);
+			ao2_cleanup(ws_client);
+			if (!ws_client) {
+				ast_ari_response_error(response, 400, "Bad Request", "external_host must be a valid websocket_client connection id.");
+				return;
+			}
+		}
+	} else {
 		external_host = ast_strdupa(args->external_host);
 		if (!ast_sockaddr_split_hostport(external_host, &host, &port, PARSE_PORT_REQUIRE)) {
-		ast_ari_response_error(response, 400, "Bad Request", "external_host must be <host>:<port>");
+			ast_ari_response_error(response, 400, "Bad Request", "external_host must be <host>:<port> for all transports other than websocket");
 			return;
 		}
+	}

 	if (ast_strlen_zero(args->format)) {
 		ast_ari_response_error(response, 400, "Bad Request", "format cannot be empty");
 		return;
 	}

-	if (ast_strlen_zero(args->encapsulation)) {
-		args->encapsulation = "rtp";
-	}
-	if (ast_strlen_zero(args->transport)) {
-		args->transport = "udp";
-	}
-	if (ast_strlen_zero(args->connection_type)) {
-		args->connection_type = "client";
-	}
 	if (ast_strlen_zero(args->direction)) {
 		args->direction = "both";
 	}
@@ -2250,6 +2344,12 @@ void ast_ari_channels_external_media(struct ast_variable *headers,
 				response, 500, "Internal Server Error",
 				"An internal error prevented this request from being handled");
 		}
+	} else if (strcasecmp(args->encapsulation, "none") == 0 && strcasecmp(args->transport, "websocket") == 0) {
+		if (external_media_websocket(args, variables, response)) {
+			ast_ari_response_error(
+				response, 500, "Internal Server Error",
+				"An internal error prevented this request from being handled");
+		}
 	} else {
 		ast_ari_response_error(
 			response, 501, "Not Implemented",
--- a/res/ari/resource_channels.h
+++ b/res/ari/resource_channels.h
@@ -834,13 +834,13 @@ struct ast_ari_channels_external_media_args {
 	const char *app;
 	/*! The "variables" key in the body object holds variable key/value pairs to set on the channel on creation. Other keys in the body object are interpreted as query parameters. Ex. { "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" } } */
 	struct ast_json *variables;
-	/*! Hostname/ip:port of external host */
+	/*! Hostname/ip:port or websocket_client connection ID of external host.  May be empty for a websocket server connection. */
 	const char *external_host;
-	/*! Payload encapsulation protocol */
+	/*! Payload encapsulation protocol.  Must be 'none' for the websocket transport. */
 	const char *encapsulation;
 	/*! Transport protocol */
 	const char *transport;
-	/*! Connection type (client/server) */
+	/*! Connection type (client/server). 'server' is only valid for the websocket transport. */
 	const char *connection_type;
 	/*! Format to encode audio in */
 	const char *format;
@@ -863,7 +863,7 @@ int ast_ari_channels_external_media_parse_body(
 /*!
 * \brief Start an External Media session.
 *
- * Create a channel to an External Media source/sink.
+ * Create a channel to an External Media source/sink.  The combination of transport and encapsulation will select one of chan_rtp(udp/rtp), chan_audiosocket(tcp/audiosocket) or chan_websocket(websocket/none) channel drivers.
 *
 * \param headers HTTP headers
 * \param args Swagger parameters
--- a/res/res_ari_channels.c
+++ b/res/res_ari_channels.c
@@ -2299,7 +2299,7 @@ static void ast_ari_channels_record_cb(
 	case 501: /* Not Implemented */
 	case 400: /* Invalid parameters */
 	case 404: /* Channel not found */
-	case 409: /* Channel is not in a Stasis application; the channel is currently bridged with other hcannels; A recording with the same name already exists on the system and can not be overwritten because it is in progress or ifExists=fail */
+	case 409: /* Channel is not in a Stasis application; the channel is currently bridged with other channels; A recording with the same name already exists on the system and can not be overwritten because it is in progress or ifExists=fail */
 	case 422: /* The format specified is unknown on this system */
 		is_valid = 1;
 		break;
--- a/res/res_http_websocket.c
+++ b/res/res_http_websocket.c
@@ -51,27 +51,21 @@
 #define MAX_PROTOCOL_BUCKETS 7

 #ifdef LOW_MEMORY
-/*! \brief Size of the pre-determined buffer for WebSocket frames */
-#define MAXIMUM_FRAME_SIZE 8192
-
 /*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
 *         payload.
 */
-#define DEFAULT_RECONSTRUCTION_CEILING 8192
+#define DEFAULT_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE

 /*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
-#define MAXIMUM_RECONSTRUCTION_CEILING 8192
+#define MAXIMUM_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE
 #else
-/*! \brief Size of the pre-determined buffer for WebSocket frames */
-#define MAXIMUM_FRAME_SIZE 65535
-
 /*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
 *         payload.
 */
-#define DEFAULT_RECONSTRUCTION_CEILING MAXIMUM_FRAME_SIZE
+#define DEFAULT_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE

 /*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
-#define MAXIMUM_RECONSTRUCTION_CEILING MAXIMUM_FRAME_SIZE
+#define MAXIMUM_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE
 #endif

 /*! \brief Maximum size of a websocket frame header
@@ -100,7 +94,7 @@ struct ast_websocket {
 	struct websocket_client *client;    /*!< Client object when connected as a client websocket */
 	char session_id[AST_UUID_STR_LEN];  /*!< The identifier for the websocket session */
 	uint16_t close_status_code;         /*!< Status code sent in a CLOSE frame upon shutdown */
-	char buf[MAXIMUM_FRAME_SIZE];	    /*!< Fixed buffer for reading data into */
+	char buf[AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE];	    /*!< Fixed buffer for reading data into */
 };

 const char *ast_websocket_type_to_str(enum ast_websocket_type type)
@@ -201,7 +195,7 @@ static void session_destroy_fn(void *obj)
 		if (session->stream) {
 			ast_iostream_close(session->stream);
 			session->stream = NULL;
-			ast_verb(2, "WebSocket connection %s '%s' closed\n", session->client ? "to" : "from",
+			ast_debug(3, "WebSocket connection %s '%s' closed\n", session->client ? "to" : "from",
 				ast_sockaddr_stringify(&session->remote_address));
 		}
 	}
@@ -279,7 +273,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_server_add_protocol2)(struct ast_websock
 	ao2_link_flags(server->protocols, protocol, OBJ_NOLOCK);
 	ao2_unlock(server->protocols);

-	ast_verb(5, "WebSocket registered sub-protocol '%s'\n", protocol->name);
+	ast_debug(1, "WebSocket registered sub-protocol '%s'\n", protocol->name);
 	ao2_ref(protocol, -1);

 	return 0;
@@ -301,7 +295,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_server_remove_protocol)(struct ast_webso
 	ao2_unlink(server->protocols, protocol);
 	ao2_ref(protocol, -1);

-	ast_verb(5, "WebSocket unregistered sub-protocol '%s'\n", name);
+	ast_debug(1, "WebSocket unregistered sub-protocol '%s'\n", name);

 	return 0;
 }
@@ -672,7 +666,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_read)(struct ast_websocket *session, cha
 		/* Now read the rest of the payload */
 		*payload = &session->buf[frame_size]; /* payload will start here, at the end of the options, if any */
 		frame_size = frame_size + (*payload_len); /* final frame size is header + optional headers + payload data */
-		if (frame_size > MAXIMUM_FRAME_SIZE) {
+		if (frame_size > AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE) {
 			ast_log(LOG_WARNING, "Cannot fit huge websocket frame of %zu bytes\n", frame_size);
 			/* The frame won't fit :-( */
 			ast_websocket_close(session, 1009);
@@ -992,7 +986,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
 		return 0;
 	}

-	ast_verb(2, "WebSocket connection from '%s' for protocol '%s' accepted using version '%d'\n", ast_sockaddr_stringify(&ser->remote_address), protocol ? : "", version);
+	ast_debug(3, "WebSocket connection from '%s' for protocol '%s' accepted using version '%d'\n", ast_sockaddr_stringify(&ser->remote_address), protocol ? : "", version);

 	/* Populate the session with all the needed details */
 	session->stream = ser->stream;
--- a/rest-api/api-docs/channels.json
+++ b/rest-api/api-docs/channels.json
@@ -1473,7 +1473,7 @@
 						},
 						{
 							"code": 409,
-							"reason": "Channel is not in a Stasis application; the channel is currently bridged with other hcannels; A recording with the same name already exists on the system and can not be overwritten because it is in progress or ifExists=fail"
+							"reason": "Channel is not in a Stasis application; the channel is currently bridged with other channels; A recording with the same name already exists on the system and can not be overwritten because it is in progress or ifExists=fail"
 						},
 						{
 							"code": 422,
@@ -1870,7 +1870,7 @@
 						"17.1.0"
 					],
 					"summary": "Start an External Media session.",
-					"notes": "Create a channel to an External Media source/sink.",
+					"notes": "Create a channel to an External Media source/sink.  The combination of transport and encapsulation will select one of chan_rtp(udp/rtp), chan_audiosocket(tcp/audiosocket) or chan_websocket(websocket/none) channel drivers.",
 					"nickname": "externalMedia",
 					"responseClass": "Channel",
 					"parameters": [
@@ -1900,15 +1900,15 @@
 						},
 						{
 							"name": "external_host",
-							"description": "Hostname/ip:port of external host",
+							"description": "Hostname/ip:port or websocket_client connection ID of external host.  May be empty for a websocket server connection.",
 							"paramType": "query",
-							"required": true,
+							"required": false,
 							"allowMultiple": false,
 							"dataType": "string"
 						},
 						{
 							"name": "encapsulation",
-							"description": "Payload encapsulation protocol",
+							"description": "Payload encapsulation protocol.  Must be 'none' for the websocket transport.",
 							"paramType": "query",
 							"required": false,
 							"allowMultiple": false,
@@ -1918,7 +1918,8 @@
 								"valueType": "LIST",
 								"values": [
 									"rtp",
-									"audiosocket"
+									"audiosocket",
+									"none"
 								]
 							}
 						},
@@ -1934,13 +1935,14 @@
 								"valueType": "LIST",
 								"values": [
 									"udp",
-									"tcp"
+									"tcp",
+									"websocket"
 								]
 							}
 						},
 						{
 							"name": "connection_type",
-							"description": "Connection type (client/server)",
+							"description": "Connection type (client/server). 'server' is only valid for the websocket transport.",
 							"paramType": "query",
 							"required": false,
 							"allowMultiple": false,
@@ -1949,7 +1951,8 @@
 							"allowableValues": {
 								"valueType": "LIST",
 								"values": [
-									"client"
+									"client",
+									"server"
 								]
 							}
 						},