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chan_sip: Allow websockets to be disabled.

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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
This commit is contained in:
Corey Farrell
2015-11-02 21:24:58 -05:00
parent 6fbffe42e1
commit 0393bd6bed
4 changed files with 30 additions and 2 deletions
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7
CHANGES
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@@ -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.

16
channels/chan_sip.c
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@@ -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();
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();
if (sip_cfg.websocket_enabled) {
ast_websocket_remove_protocol("sip", sip_websocket_callback);
}
network_change_stasis_unsubscribe();
acl_change_event_stasis_unsubscribe();

1
channels/sip/include/sip.h
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@@ -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;

4
configs/samples/sip.conf.sample
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@@ -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