chan_sip: Allow websockets to be disabled.

This patch adds a new setting "websockets_enabled" to sip.conf.
Setting this to false allows chan_sip to be used without causing
conflicts with res_pjsip_transport_websocket.

ASTERISK-24106 #close
Reported by: Andrew Nagy

Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7

CHANGES

@@ -200,6 +200,13 @@ Queue
 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
 ------------------------------------------------------------------------------
 
+chan_sip
+------------------
+ * The websockets_enabled option has been added to the general section of
+   sip.conf.  The option is enabled by default to match the previous behavior.
+   The option should be disabled when using res_pjsip_transport_websockets to
+   ensure chan_sip will not conflict with PJSIP websockets.
+
 Dialplan Functions
 ------------------
  * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.

channels/chan_sip.c

@@ -31265,6 +31265,7 @@ static int reload_config(enum channelreloadreason reason)
 	int bindport = 0;
 	int acl_change_subscription_needed = 0;
 	int min_subexpiry_set = 0, max_subexpiry_set = 0;
+	int websocket_was_enabled = sip_cfg.websocket_enabled;
 
 	run_start = time(0);
 	ast_unload_realtime("sipregs");
@@ -32068,6 +32069,8 @@ static int reload_config(enum channelreloadreason reason)
 				ast_log(LOG_WARNING, "'%s' is not a valid websocket_write_timeout value at line %d. Using default '%d'.\n", v->value, v->lineno, AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT);
 				sip_cfg.websocket_write_timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
 			}
+		} else if (!strcasecmp(v->name, "websocket_enabled")) {
+			sip_cfg.websocket_enabled = ast_true(v->value);
 		}
 	}
 
@@ -32413,6 +32416,15 @@ static int reload_config(enum channelreloadreason reason)
 		notify_types = NULL;
 	}
 
+	/* If the module is loading it's not time to enable websockets yet. */
+	if (reason != CHANNEL_MODULE_LOAD && websocket_was_enabled != sip_cfg.websocket_enabled) {
+		if (sip_cfg.websocket_enabled) {
+			ast_websocket_add_protocol("sip", sip_websocket_callback);
+		} else {
+			ast_websocket_remove_protocol("sip", sip_websocket_callback);
+		}
+	}
+
 	run_end = time(0);
 	ast_debug(4, "SIP reload_config done...Runtime= %d sec\n", (int)(run_end-run_start));
 
@@ -34594,7 +34606,9 @@ static int load_module(void)
 	sip_register_tests();
 	network_change_stasis_subscribe();
 
+	if (sip_cfg.websocket_enabled) {
 		ast_websocket_add_protocol("sip", sip_websocket_callback);
+	}
 
 	return AST_MODULE_LOAD_SUCCESS;
 }
@@ -34609,7 +34623,9 @@ static int unload_module(void)
 
 	ast_sip_api_provider_unregister();
 
+	if (sip_cfg.websocket_enabled) {
 		ast_websocket_remove_protocol("sip", sip_websocket_callback);
+	}
 
 	network_change_stasis_unsubscribe();
 	acl_change_event_stasis_unsubscribe();

channels/sip/include/sip.h

@@ -774,6 +774,7 @@ struct sip_settings {
 	int tcp_enabled;
 	int default_max_forwards;    /*!< Default max forwards (SIP Anti-loop) */
 	int websocket_write_timeout; /*!< Socket write timeout for websocket transports, in ms */
+	int websocket_enabled;       /*!< Are websockets enabled? */
 };
 
 struct ast_websocket;

configs/samples/sip.conf.sample

@@ -232,6 +232,10 @@ tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0
 				; unauthenticated sessions that will be allowed
                                 ; to connect at any given time. (default: 100)
 
+;websocket_enabled = true       ; Set to false to prevent chan_sip from listening to websockets.  This
+                                ; is neeeded when using chan_sip and res_pjsip_transport_websockets on
+                                ; the same system.
+
 ;websocket_write_timeout = 100  ; Default write timeout to set on websocket transports.
                                 ; This value may need to be adjusted for connections where
                                 ; Asterisk must write a substantial amount of data and the