diff --git a/CHANGES b/CHANGES
index 044605cc52108500b48260d72c7ce847aed53a64..8c476e43329868ac395c0a0dda764d8f586a54e7 100644
--- 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
--- 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
--- 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
--- 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