[asterisk-commits] res/res pjsip dlg options: Add a module to handle in-dialog ... (asterisk[certified/13.1])

[SVN commits to the Asterisk project]

Mark Michelson has submitted this change and it was merged.

Change subject: res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests
......................................................................


res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests

This patch adds a new session supplement that handles in-dialog OPTIONS
requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup
for the OPTIONS request would already have been done by the time the
session supplement receives the inbound request.

ASTERISK-24862 #close
Reported by: yaron nahum
patches:
  res_pjsip_dlg_options.c submitted by yaron nahum (License 6676)

Change-Id: Iefc901a7c5c88d9d4b853188f85092d9eb7b6ada
---
M UPGRADE.txt
A res/res_pjsip_dlg_options.c
2 files changed, 112 insertions(+), 1 deletion(-)

Approvals:
  Mark Michelson: Looks good to me, approved; Verified
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/UPGRADE.txt b/UPGRADE.txt
index acec1ab..da1a857 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -21,9 +21,13 @@
 === UPGRADE-12.txt  -- Upgrade info for 11 to 12
 ===========================================================
 
-
 From 13.1-cert2 to 13.1-cert3:
 
+res_pjsip_dlg_options:
+ - A new module, this handles OPTIONS requests sent in-dialog. This module
+   should have no adverse effects for those upgrading; this note merely
+   serves as an indication that a new module exists.
+
 Source Control:
  - Asterisk has moved from Subversion to Git. As a result, several changes
    were required in functionality. These are listed individually in the
diff --git a/res/res_pjsip_dlg_options.c b/res/res_pjsip_dlg_options.c
new file mode 100644
index 0000000..54c9f86
--- /dev/null
+++ b/res/res_pjsip_dlg_options.c
@@ -0,0 +1,107 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2015, Digium, Inc.
+ *
+ * Yaron Nahum <nachum.yaron 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_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+#include <pjlib.h>
+
+#include "asterisk/module.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+
+#define DEFAULT_LANGUAGE "en"
+#define DEFAULT_ENCODING "text/plain"
+
+static int options_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
+{
+	pjsip_tx_data *tdata;
+        pj_status_t status;
+	const pjsip_hdr *hdr;
+	pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
+
+	status = pjsip_dlg_create_response(session->inv_session->dlg, rdata, 200, NULL,&tdata);
+	if (status != PJ_SUCCESS) {
+		ast_log(LOG_ERROR, "Unable to create response (%d)\n", status);
+		return status;
+	}
+
+	/* Add appropriate headers */
+	if ((hdr = pjsip_endpt_get_capability(endpt, PJSIP_H_ACCEPT, NULL))) {
+		pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)pjsip_hdr_clone(tdata->pool, hdr));
+	}
+	if ((hdr = pjsip_endpt_get_capability(endpt, PJSIP_H_ALLOW, NULL))) {
+		pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)pjsip_hdr_clone(tdata->pool, hdr));
+	}
+	if ((hdr = pjsip_endpt_get_capability(endpt, PJSIP_H_SUPPORTED, NULL))) {
+		pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)pjsip_hdr_clone(tdata->pool, hdr));
+	}
+
+	/*
+	 * XXX TODO: pjsip doesn't care a lot about either of these headers -
+	 * while it provides specific methods to create them, they are defined
+	 * to be the standard string header creation. We never did add them
+	 * in chan_sip, although RFC 3261 says they SHOULD. Hard coded here.
+	*/
+	ast_sip_add_header(tdata, "Accept-Encoding", DEFAULT_ENCODING);
+	ast_sip_add_header(tdata, "Accept-Language", DEFAULT_LANGUAGE);
+
+	status = pjsip_dlg_send_response(session->inv_session->dlg, pjsip_rdata_get_tsx(rdata), tdata);
+	if (status != PJ_SUCCESS) {
+		ast_log(LOG_ERROR, "Unable to send response (%d)\n", status);
+	}
+
+	return status;
+}
+
+static struct ast_sip_session_supplement  dlg_options_supplement = {
+	.method = "OPTIONS",
+	.incoming_request = options_incoming_request,
+};
+
+static int load_module(void)
+{
+	CHECK_PJSIP_MODULE_LOADED();
+
+	if (ast_sip_session_register_supplement(&dlg_options_supplement)) {
+		return AST_MODULE_LOAD_DECLINE;
+	}
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_sip_session_unregister_supplement(&dlg_options_supplement);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP OPTIONS in dialog handler",
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_APP_DEPEND,
+);

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: Iefc901a7c5c88d9d4b853188f85092d9eb7b6ada
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.1
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>