[asterisk-commits] mmichelson: branch mmichelson/sip_options r393970 - in /team/mmichelson/sip_o...

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Wed Jul 10 09:16:06 2013
New Revision: 393970

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=393970
Log:
Add 'rtpengine' option for SIP endpoints.


Modified:
    team/mmichelson/sip_options/include/asterisk/res_sip.h
    team/mmichelson/sip_options/res/res_sip/sip_configuration.c
    team/mmichelson/sip_options/res/res_sip_sdp_rtp.c

Modified: team/mmichelson/sip_options/include/asterisk/res_sip.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_options/include/asterisk/res_sip.h?view=diff&rev=393970&r1=393969&r2=393970
==============================================================================
--- team/mmichelson/sip_options/include/asterisk/res_sip.h (original)
+++ team/mmichelson/sip_options/include/asterisk/res_sip.h Wed Jul 10 09:16:06 2013
@@ -323,6 +323,8 @@
 		AST_STRING_FIELD(external_media_address);
 		/*! Configured voicemail boxes for this endpoint. Used for MWI */
 		AST_STRING_FIELD(mailboxes);
+		/*! Configured RTP engine for this endpoint. */
+		AST_STRING_FIELD(rtp_engine);
 	);
 	/*! Identification information for this endpoint */
 	struct ast_party_id id;

Modified: team/mmichelson/sip_options/res/res_sip/sip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_options/res/res_sip/sip_configuration.c?view=diff&rev=393970&r1=393969&r2=393970
==============================================================================
--- team/mmichelson/sip_options/res/res_sip/sip_configuration.c (original)
+++ team/mmichelson/sip_options/res/res_sip/sip_configuration.c Wed Jul 10 09:16:06 2013
@@ -658,6 +658,7 @@
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "namedcallgroup", "", named_groups_handler, NULL, 0, 0);
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "namedpickupgroup", "", named_groups_handler, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "devicestate_busy_at", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, devicestate_busy_at));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtpengine", "asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, rtp_engine));
 
 	if (ast_sip_initialize_sorcery_transport(sip_sorcery)) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");

Modified: team/mmichelson/sip_options/res/res_sip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_options/res/res_sip_sdp_rtp.c?view=diff&rev=393970&r1=393969&r2=393970
==============================================================================
--- team/mmichelson/sip_options/res/res_sip_sdp_rtp.c (original)
+++ team/mmichelson/sip_options/res/res_sip_sdp_rtp.c Wed Jul 10 09:16:06 2013
@@ -108,7 +108,8 @@
 {
 	struct ast_rtp_engine_ice *ice;
 
-	if (!(session_media->rtp = ast_rtp_instance_new("asterisk", sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
+	if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->rtp_engine, sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
+		ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->rtp_engine);
 		return -1;
 	}