diff options
author | Matthew Jordan <mjordan@digium.com> | 2015-04-09 15:42:16 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2015-04-09 15:42:16 +0000 |
commit | 2679d0100af28d47bd320b1e7045bde707517789 (patch) | |
tree | 1e108be03f07c665924515f30ed07a7126815aea /res | |
parent | 6ba6e3dffd57f1c62c0c7f9a9030c1f41b5a6350 (diff) |
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip_dlg_options.c | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/res/res_pjsip_dlg_options.c b/res/res_pjsip_dlg_options.c new file mode 100644 index 000000000..54c9f860f --- /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@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, +); |