[Asterisk-code-review] chan sip deprecation. (asterisk[master])
Corey Farrell
asteriskteam at digium.com
Wed Oct 10 07:26:13 CDT 2018
Corey Farrell has uploaded this change for review. ( https://gerrit.asterisk.org/10460
Change subject: chan_sip deprecation.
......................................................................
chan_sip deprecation.
This officially deprecates chan_sip in Asterisk 17+. A warning is
printed upon startup or module load to tell users that they should
consider migrating. chan_sip is still built by default but the default
modules.conf skips loading it at startup.
Very important to note we are not scheduling a time where chan_sip will
be removed. The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.
Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93
---
M CHANGES
M UPGRADE.txt
M channels/chan_sip.c
M channels/sip/config_parser.c
M channels/sip/dialplan_functions.c
M channels/sip/reqresp_parser.c
M channels/sip/route.c
M channels/sip/security_events.c
M channels/sip/utils.c
M configs/samples/modules.conf.sample
10 files changed, 68 insertions(+), 10 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/60/10460/1
diff --git a/CHANGES b/CHANGES
index bd1a7a7..5bfdc54 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,17 @@
==============================================================================
------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16 to Asterisk 17 --------------------
+------------------------------------------------------------------------------
+
+chan_sip
+------------------
+ * The chan_sip module is now deprecated, users should migrate to the
+ replacement module chan_pjsip. See guides at the Asterisk Wiki:
+ https://wiki.asterisk.org/wiki/x/tAHOAQ
+ https://wiki.asterisk.org/wiki/x/hYCLAQ
+
+------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
------------------------------------------------------------------------------
diff --git a/UPGRADE.txt b/UPGRADE.txt
index b7bbf3b..937182d 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -25,3 +25,11 @@
=== UPGRADE-16.txt -- Upgrade info for 15 to 16
===========================================================
+New in 17.0.0:
+
+chan_sip:
+ - The chan_sip module is now deprecated, users should migrate to the
+ replacement module chan_pjsip. See guides at the Asterisk Wiki:
+ https://wiki.asterisk.org/wiki/x/tAHOAQ
+ https://wiki.asterisk.org/wiki/x/hYCLAQ
+
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 42362b8..6c681a1 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -176,7 +176,7 @@
/*** MODULEINFO
<use type="module">res_crypto</use>
<use type="module">res_http_websocket</use>
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
/*! \page sip_session_timers SIP Session Timers in Asterisk Chan_sip
@@ -35321,6 +35321,37 @@
.sipinfo_send = sipinfo_send,
};
+static void deprecation_notice(void)
+{
+ ast_log(LOG_WARNING, "chan_sip has no official maintainer and is deprecated. Migration to\n");
+ ast_log(LOG_WARNING, "chan_pjsip is recommended. See guides at the Asterisk Wiki:\n");
+ ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/x/tAHOAQ\n");
+ ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/x/hYCLAQ\n");
+}
+
+/*! \brief Event callback which indicates we're fully booted */
+static void startup_event_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message)
+{
+ struct ast_json_payload *payload;
+ const char *type;
+
+ if (stasis_message_type(message) != ast_manager_get_generic_type()) {
+ return;
+ }
+
+ payload = stasis_message_data(message);
+ type = ast_json_string_get(ast_json_object_get(payload->json, "type"));
+
+ if (strcmp(type, "FullyBooted")) {
+ return;
+ }
+
+ deprecation_notice();
+
+ stasis_unsubscribe(sub);
+}
+
+
static int unload_module(void);
/*!
@@ -35527,6 +35558,12 @@
ast_websocket_add_protocol("sip", sip_websocket_callback);
}
+ if (ast_fully_booted) {
+ deprecation_notice();
+ } else {
+ stasis_subscribe_pool(ast_manager_get_topic(), startup_event_cb, NULL);
+ }
+
return AST_MODULE_LOAD_SUCCESS;
}
@@ -35744,7 +35781,7 @@
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
- .support_level = AST_MODULE_SUPPORT_CORE,
+ .support_level = AST_MODULE_SUPPORT_DEPRECATED,
.load = load_module,
.unload = unload_module,
.reload = reload,
diff --git a/channels/sip/config_parser.c b/channels/sip/config_parser.c
index 211f600..64cb66a 100644
--- a/channels/sip/config_parser.c
+++ b/channels/sip/config_parser.c
@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
#include "asterisk.h"
diff --git a/channels/sip/dialplan_functions.c b/channels/sip/dialplan_functions.c
index 09804ce..7c34fc9 100644
--- a/channels/sip/dialplan_functions.c
+++ b/channels/sip/dialplan_functions.c
@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
/*** DOCUMENTATION
diff --git a/channels/sip/reqresp_parser.c b/channels/sip/reqresp_parser.c
index 4d91446..b43bed6 100644
--- a/channels/sip/reqresp_parser.c
+++ b/channels/sip/reqresp_parser.c
@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
#include "asterisk.h"
diff --git a/channels/sip/route.c b/channels/sip/route.c
index 584b46b..916c3af 100644
--- a/channels/sip/route.c
+++ b/channels/sip/route.c
@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
#include "asterisk.h"
diff --git a/channels/sip/security_events.c b/channels/sip/security_events.c
index 86ba413..eabc904 100644
--- a/channels/sip/security_events.c
+++ b/channels/sip/security_events.c
@@ -25,7 +25,7 @@
*/
/*** MODULEINFO
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
#include "asterisk.h"
diff --git a/channels/sip/utils.c b/channels/sip/utils.c
index 45a4128..d53316c 100644
--- a/channels/sip/utils.c
+++ b/channels/sip/utils.c
@@ -22,7 +22,7 @@
*/
/*** MODULEINFO
- <support_level>extended</support_level>
+ <support_level>deprecated</support_level>
***/
#include "asterisk.h"
diff --git a/configs/samples/modules.conf.sample b/configs/samples/modules.conf.sample
index 179e9ef..df6784f 100644
--- a/configs/samples/modules.conf.sample
+++ b/configs/samples/modules.conf.sample
@@ -25,7 +25,7 @@
; the "require" keyword. Asterisk will exit with a status code of 2
; if a required module does not load.
;
-; require = chan_sip.so
+; require = chan_pjsip.so
; If you want you can combine with preload
; preload-require = res_odbc.so
;
@@ -41,4 +41,6 @@
noload => res_hep.so
noload => res_hep_pjsip.so
noload => res_hep_rtcp.so
-;
+
+; Do not load chan_sip by default, it may conflict with res_pjsip.
+noload => chan_sip.so
--
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93
Gerrit-Change-Number: 10460
Gerrit-PatchSet: 1
Gerrit-Owner: Corey Farrell <git at cfware.com>
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