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authorJoshua Colp <jcolp@digium.com>2014-12-12 13:06:24 +0000
committerJoshua Colp <jcolp@digium.com>2014-12-12 13:06:24 +0000
commit74d43977cf1ec43b9d5d8f84cc64592fdbd11115 (patch)
treee49497e449926fdbb08abb648437bd40419c94da /include/asterisk/res_pjsip_session.h
parent8d384f38254142b6b41d335deebaf03fb8a8038b (diff)
res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.
Given the scenario where a PJSIP channel is in a native RTP bridge with direct media and the channel is then hung up the code will currently re-INVITE the channel back to Asterisk and send a BYE at the same time. Many SIP implementations dislike this greatly. This change makes it so that if a re-INVITE transaction is in progress the BYE is queued to occur after the completion of the transaction (be it through normal means or a timeout). Review: https://reviewboard.asterisk.org/r/4248/ ........ Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk/res_pjsip_session.h')
-rw-r--r--include/asterisk/res_pjsip_session.h8
1 files changed, 8 insertions, 0 deletions
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 9a133fcb3..990c9fe6f 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -436,6 +436,14 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
struct ast_format_cap *req_caps);
/*!
+ * \brief Terminate a session and, if possible, send the provided response code
+ *
+ * \param session The session to terminate
+ * \param response The response code to use for termination if possible
+ */
+void ast_sip_session_terminate(struct ast_sip_session *session, int response);
+
+/*!
* \brief Defer local termination of a session until remote side terminates, or an amount of time passes
*
* \param session The session to defer termination on