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

diff --git a/CHANGES b/CHANGES
index 044605cc52108500b48260d72c7ce847aed53a64..8c476e43329868ac395c0a0dda764d8f586a54e7 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,13 @@
 --- 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.

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f28296627389e16c10ef3d5ffb927dcb232ae623..acb7d535fb8a106157598ca116abd6196a14347e 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -31261,6 +31261,7 @@ 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");
@@ -32047,6 +32048,8 @@ 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);
                }
        }
 
@@ -32392,6 +32395,15 @@ 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));
 
@@ -34573,7 +34585,9 @@ static int load_module(void)
        sip_register_tests();
        network_change_stasis_subscribe();
 
-       ast_websocket_add_protocol("sip", sip_websocket_callback);
+       if (sip_cfg.websocket_enabled) {
+               ast_websocket_add_protocol("sip", sip_websocket_callback);
+       }
 
        return AST_MODULE_LOAD_SUCCESS;
 }
@@ -34588,7 +34602,9 @@ static int unload_module(void)
 
        ast_sip_api_provider_unregister();
 
-       ast_websocket_remove_protocol("sip", sip_websocket_callback);
+       if (sip_cfg.websocket_enabled) {
+               ast_websocket_remove_protocol("sip", sip_websocket_callback);
+       }
 
        network_change_stasis_unsubscribe();
        acl_change_event_stasis_unsubscribe();

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 3ed3e8a33caf63fd9e370b88ef1d79d47db3b649..82f208c77468789bf47c20e0bb55a88b94f73585 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/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;

diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index 44d2d4352f10d9250a184c2662e382f60cd206a4..a24ab30a678228220b4cd6d8b07b885bba9bfc16 100644 (file)
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -229,6 +229,10 @@ 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