[Asterisk-code-review] res pjsip session: Add cleanup to ast sip session terminate (asterisk[14])
Jenkins2
asteriskteam at digium.com
Thu Apr 27 17:09:01 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5547 )
Change subject: res_pjsip_session: Add cleanup to ast_sip_session_terminate
......................................................................
res_pjsip_session: Add cleanup to ast_sip_session_terminate
If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
---
M channels/chan_pjsip.c
M include/asterisk/res_pjsip_session.h
M res/res_pjsip_session.c
3 files changed, 47 insertions(+), 5 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Mark Michelson: Looks good to me, approved
Jenkins2: Approved for Submit
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index e8e3666..851c913 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -2054,11 +2054,16 @@
struct ast_sip_session *session = channel->session;
int cause = h_data->cause;
- ast_sip_session_terminate(session, cause);
+ /*
+ * It's possible that session_terminate might cause the session to be destroyed
+ * immediately so we need to keep a reference to it so we can NULL session->channel
+ * afterwards.
+ */
+ ast_sip_session_terminate(ao2_bump(session), cause);
clear_session_and_channel(session, ast, pvt);
+ ao2_cleanup(session);
ao2_cleanup(channel);
ao2_cleanup(h_data);
-
return 0;
}
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index d4d3f70..10e55f1 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -459,6 +459,10 @@
*
* \param session The session to terminate
* \param response The response code to use for termination if possible
+ *
+ * \warning Calling this function MAY cause the last session reference to be
+ * released and the session destructor to be called. If you need to do something
+ * with session after this call, be sure to bump the ref count before calling terminate.
*/
void ast_sip_session_terminate(struct ast_sip_session *session, int response);
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index d98bc0a..59bd9d7 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1300,9 +1300,19 @@
struct ast_sip_session *session = obj;
struct ast_sip_session_supplement *supplement;
struct ast_sip_session_delayed_request *delay;
+ const char *endpoint_name = session->endpoint ?
+ ast_sorcery_object_get_id(session->endpoint) : "<none>";
- ast_debug(3, "Destroying SIP session with endpoint %s\n",
- session->endpoint ? ast_sorcery_object_get_id(session->endpoint) : "<none>");
+ ast_debug(3, "Destroying SIP session with endpoint %s\n", endpoint_name);
+
+ ast_test_suite_event_notify("SESSION_DESTROYING",
+ "Endpoint: %s\r\n"
+ "AOR: %s\r\n"
+ "Contact: %s"
+ , endpoint_name
+ , session->aor ? ast_sorcery_object_get_id(session->aor) : "<none>"
+ , session->contact ? ast_sorcery_object_get_id(session->contact) : "<none>"
+ );
while ((supplement = AST_LIST_REMOVE_HEAD(&session->supplements, next))) {
if (supplement->session_destroy) {
@@ -1331,6 +1341,8 @@
if (session->inv_session) {
pjsip_dlg_dec_session(session->inv_session->dlg, &session_module);
}
+
+ ast_test_suite_event_notify("SESSION_DESTROYED", "Endpoint: %s", endpoint_name);
}
static int add_supplements(struct ast_sip_session *session)
@@ -1791,6 +1803,9 @@
return ret_session;
}
+static int session_end(void *vsession);
+static int session_end_completion(void *vsession);
+
void ast_sip_session_terminate(struct ast_sip_session *session, int response)
{
pj_status_t status;
@@ -1807,7 +1822,25 @@
switch (session->inv_session->state) {
case PJSIP_INV_STATE_NULL:
- pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
+ if (!session->inv_session->invite_tsx) {
+ /*
+ * Normally, it's pjproject's transaction cleanup that ultimately causes the
+ * final session reference to be released but if both STATE and invite_tsx are NULL,
+ * we never created a transaction in the first place. In this case, we need to
+ * do the cleanup ourselves.
+ */
+ /* Transfer the inv_session session reference to the session_end_task */
+ session->inv_session->mod_data[session_module.id] = NULL;
+ pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
+ session_end(session);
+ /*
+ * session_end_completion will cleanup the final session reference unless
+ * ast_sip_session_terminate's caller is holding one.
+ */
+ session_end_completion(session);
+ } else {
+ pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
+ }
break;
case PJSIP_INV_STATE_CONFIRMED:
if (session->inv_session->invite_tsx) {
--
To view, visit https://gerrit.asterisk.org/5547
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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