[asterisk-commits] res pjsip empty info: Respond to empty SIP INFO packets (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 19 14:46:16 CDT 2016
Joshua Colp has submitted this change and it was merged.
Change subject: res_pjsip_empty_info: Respond to empty SIP INFO packets
......................................................................
res_pjsip_empty_info: Respond to empty SIP INFO packets
Some SBCs require responses to empty SIP INFO packets
after establishing call via INVITE, if not responded to
they may drop your call after unspecified timeout of X minutes.
They are identified by having no Content-Type, check for this
and respond with 200 - OK message.
ASTERISK-24986 #close
Reported-by: Ilya Trikoz, Federico Santulli
Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0
---
M CHANGES
M res/res_pjsip_dtmf_info.c
A res/res_pjsip_empty_info.c
M res/res_pjsip_one_touch_record_info.c
4 files changed, 105 insertions(+), 7 deletions(-)
Approvals:
George Joseph: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, approved; Verified
diff --git a/CHANGES b/CHANGES
index 91f5ce5..a22c0cd 100644
--- a/CHANGES
+++ b/CHANGES
@@ -252,6 +252,12 @@
will be used instead. The new SIP resolver provides NAPTR support, improved
SRV support, and AAAA record support.
+res_pjsip_info_empty
+--------------------
+ * A new module that can respond to empty Content-Type INFO packets during call.
+ Some SBCs will terminate a call if their empty INFO packets are not responded
+ to within a predefined time.
+
res_pjsip_outbound_registration
-------------------------------
* A new 'fatal_retry_interval' option has been added to outbound registration.
@@ -264,6 +270,7 @@
* Added a new multi_user option that when set to 'yes' allows a given configuration
to be used for multiple users.
+
CEL Backends
------------------
diff --git a/res/res_pjsip_dtmf_info.c b/res/res_pjsip_dtmf_info.c
index 47ccd1a..e534f3e 100644
--- a/res/res_pjsip_dtmf_info.c
+++ b/res/res_pjsip_dtmf_info.c
@@ -103,13 +103,13 @@
if (!body || !body->len) {
/* need to return 200 OK on empty body */
send_response(session, rdata, 200);
- return 0;
+ return 1;
}
res = body->print_body(body, buf, body->len);
if (res < 0) {
send_response(session, rdata, 500);
- return 0;
+ return 1;
}
buf[res] = '\0';
@@ -150,11 +150,12 @@
}
send_response(session, rdata, event ? 200 : 500);
- return event ? 0 : -1;
+ return 1;
}
static struct ast_sip_session_supplement dtmf_info_supplement = {
.method = "INFO",
+ .priority = AST_SIP_SUPPLEMENT_PRIORITY_FIRST,
.incoming_request = dtmf_info_incoming_request,
};
diff --git a/res/res_pjsip_empty_info.c b/res/res_pjsip_empty_info.c
new file mode 100644
index 0000000..09109ba
--- /dev/null
+++ b/res/res_pjsip_empty_info.c
@@ -0,0 +1,89 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Digium, Inc.
+ *
+ * Bradley Latus <brad.latus at gmail.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*** MODULEINFO
+ <depend>pjproject</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_REGISTER_FILE()
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+#include "asterisk/module.h"
+
+static void send_response(struct ast_sip_session *session,
+ struct pjsip_rx_data *rdata, int code)
+{
+ pjsip_tx_data *tdata;
+ pjsip_dialog *dlg = session->inv_session->dlg;
+
+ if (pjsip_dlg_create_response(dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
+ struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+ pjsip_dlg_send_response(dlg, tsx, tdata);
+ }
+}
+
+static int empty_info_incoming_request(struct ast_sip_session *session,
+ struct pjsip_rx_data *rdata)
+{
+ if (!rdata->msg_info.ctype) {
+ /* Need to return 200 OK on empty body */
+ /* Some SBCs use empty INFO as a KEEPALIVE */
+ send_response(session, rdata, 200);
+ return 1;
+ }
+
+ /* Let another module respond */
+ return 0;
+
+}
+
+static struct ast_sip_session_supplement empty_info_supplement = {
+ .method = "INFO",
+ .priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST,
+ .incoming_request = empty_info_incoming_request,
+};
+
+static int load_module(void)
+{
+ CHECK_PJSIP_SESSION_MODULE_LOADED();
+
+ ast_sip_session_register_supplement(&empty_info_supplement);
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+ ast_sip_session_unregister_supplement(&empty_info_supplement);
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Empty INFO Support",
+ .support_level = AST_MODULE_SUPPORT_CORE,
+ .load = load_module,
+ .unload = unload_module,
+ .load_pri = AST_MODPRI_APP_DEPEND,
+);
diff --git a/res/res_pjsip_one_touch_record_info.c b/res/res_pjsip_one_touch_record_info.c
index 30dfcaa..2d53fd4 100644
--- a/res/res_pjsip_one_touch_record_info.c
+++ b/res/res_pjsip_one_touch_record_info.c
@@ -72,13 +72,13 @@
if (!session->channel) {
send_response(session, 481, rdata);
- return 0;
+ return 1;
}
/* Is this endpoint configured with One Touch Recording? */
if (!session->endpoint->info.recording.enabled || ast_strlen_zero(feature)) {
send_response(session, 403, rdata);
- return 0;
+ return 1;
}
ast_channel_lock(session->channel);
@@ -87,7 +87,7 @@
if (feature_res || ast_strlen_zero(feature_code)) {
send_response(session, 403, rdata);
- return 0;
+ return 1;
}
for (digit = feature_code; *digit; ++digit) {
@@ -97,11 +97,12 @@
send_response(session, 200, rdata);
- return 0;
+ return 1;
}
static struct ast_sip_session_supplement info_supplement = {
.method = "INFO",
+ .priority = AST_SIP_SUPPLEMENT_PRIORITY_FIRST,
.incoming_request = handle_incoming_request,
};
--
To view, visit https://gerrit.asterisk.org/2710
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0
Gerrit-PatchSet: 8
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: snuffy <snuffy22 at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: snuffy <snuffy22 at gmail.com>
More information about the asterisk-commits
mailing list