Residual changes for Asterisk v10 branch from ASTERISK-18747.

Residual changes for Asterisk v10 branch from ASTERISK-18747 after
https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs
callid hash key change fix.

* Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from
dialogs_rtpcheck.  This is an optimization to avoid an unneeded
lock/unlock and object search when using ao2_unlink.

* Prevent crash in check_rtp_timeout() if dialog->rtp is NULL.

Review: https://reviewboard.asterisk.org/r/1557/
........

Merged revisions 344004 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett 2011-11-08 22:14:38 +00:00
parent bc370b5462
commit 9537f22c54
1 changed files with 64 additions and 47 deletions

View File

@ -1078,21 +1078,24 @@ static void destroy_escs(void)
}
}
/*! \brief
/*!
* \details
* Here we implement the container for dialogs which are in the
* dialog_needdestroy state to iterate only through the dialogs
* unlink them instead of iterate through all dialogs
*/
struct ao2_container *dialogs_needdestroy;
/*! \brief
/*!
* \details
* Here we implement the container for dialogs which have rtp
* traffic and rtptimeout, rtpholdtimeout or rtpkeepalive
* set. We use this container instead the whole dialog list.
*/
struct ao2_container *dialogs_rtpcheck;
/*! \brief
/*!
* \details
* Here we implement the container for dialogs (sip_pvt), defining
* generic wrapper functions to ease the transition from the current
* implementation (a single linked list) to a different container.
@ -1341,7 +1344,7 @@ static void add_realm_authentication(struct sip_auth_container **credentials, co
static struct sip_auth *find_realm_authentication(struct sip_auth_container *credentials, const char *realm);
/*--- Misc functions */
static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
static int check_rtp_timeout(struct sip_pvt *dialog, time_t t);
static int reload_config(enum channelreloadreason reason);
static void add_diversion_header(struct sip_request *req, struct sip_pvt *pvt);
static int expire_register(const void *data);
@ -2948,15 +2951,6 @@ static void ref_proxy(struct sip_pvt *pvt, struct sip_proxy *proxy)
}
}
/*!
* \brief Unlink a dialog from the dialogs_checkrtp container
*/
static void *dialog_unlink_rtpcheck(struct sip_pvt *dialog)
{
ao2_t_unlink(dialogs_rtpcheck, dialog, "unlinking dialog_rtpcheck via ao2_unlink");
return NULL;
}
/*!
* \brief Unlink a dialog from the dialogs container, as well as any other places
* that it may be currently stored.
@ -3077,11 +3071,11 @@ static inline void pvt_set_needdestroy(struct sip_pvt *pvt, const char *reason)
if (pvt->final_destruction_scheduled) {
return; /* This is already scheduled for final destruction, let the scheduler take care of it. */
}
if(pvt->needdestroy != 1) {
append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
if (!pvt->needdestroy) {
pvt->needdestroy = 1;
ao2_t_link(dialogs_needdestroy, pvt, "link pvt into dialogs_needdestroy container");
}
append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
pvt->needdestroy = 1;
}
/*! \brief Initialize the initital request packet in the pvt structure.
@ -6268,7 +6262,6 @@ static int sip_hangup(struct ast_channel *ast)
ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
p->needdestroy = 0;
p->owner->tech_pvt = dialog_unref(p->owner->tech_pvt, "unref p->owner->tech_pvt");
sip_pvt_lock(p);
p->owner = NULL; /* Owner will be gone after we return, so take it away */
@ -11731,9 +11724,14 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
/* we unlink this dialog and link again into the dialogs_rtpcheck container to doesnt add it twice */
/*
* We unlink this dialog and link again into the
* dialogs_rtpcheck container so its not in there twice.
*/
ao2_lock(dialogs_rtpcheck);
ao2_t_unlink(dialogs_rtpcheck, p, "unlink pvt into dialogs_rtpcheck container");
ao2_t_link(dialogs_rtpcheck, p, "link pvt into dialogs_rtpcheck container");
ao2_unlock(dialogs_rtpcheck);
ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmpcap));
@ -17096,28 +17094,33 @@ static void cleanup_stale_contexts(char *new, char *old)
}
}
/*! \brief Check RTP Timeout on dialogs
/*!
* \brief Check RTP Timeout on dialogs
*
* \details This is used with ao2_callback to check rtptimeout
* rtponholdtimeout and send rtpkeepalive packets
* rtponholdtimeout and send rtpkeepalive packets.
*
* \return CMP_MATCH for items to be unlinked from dialogs_rtpcheck.
*/
static int dialog_checkrtp_cb(void *dialogobj, void *arg, int flags)
{
struct sip_pvt *dialog = dialogobj;
time_t *t = arg;
int match_status;
if (sip_pvt_trylock(dialog)) {
return 0;
}
if (dialog->rtp || dialog->vrtp) {
check_rtp_timeout(dialog, *t);
match_status = check_rtp_timeout(dialog, *t);
} else {
/* Dialog has no active RTP or VRTP. unlink it from the checkrtp container */
dialog_unlink_rtpcheck(dialog);
/* Dialog has no active RTP or VRTP. unlink it from dialogs_rtpcheck. */
match_status = CMP_MATCH;
}
sip_pvt_unlock(dialog);
return 0;
return match_status;
}
/*!
@ -17140,7 +17143,6 @@ static int dialog_needdestroy(void *dialogobj, void *arg, int flags)
return 0;
}
/* If we have sessions that needs to be destroyed, do it now */
/* Check if we have outstanding requests not responsed to or an active call
- if that's the case, wait with destruction */
@ -26002,33 +26004,42 @@ static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only)
return 0;
}
/*! \brief helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked */
static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
/*!
* \brief helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked
*
* \return CMP_MATCH for items to be unlinked from dialogs_rtpcheck.
*/
static int check_rtp_timeout(struct sip_pvt *dialog, time_t t)
{
int timeout;
int hold_timeout;
int keepalive;
if (!dialog->rtp) {
/*
* We have no RTP. Since we don't do much with video RTP for
* now, stop checking this dialog.
*/
return CMP_MATCH;
}
/* If we have no active owner, no need to check timers */
if (!dialog->owner) {
dialog_unlink_rtpcheck(dialog);
return;
return CMP_MATCH;
}
/* If the call is redirected outside Asterisk, no need to check timers */
/* If the call is redirected outside Asterisk, no need to check timers */
if (!ast_sockaddr_isnull(&dialog->redirip)) {
dialog_unlink_rtpcheck(dialog);
return;
return CMP_MATCH;
}
/* If the call is involved in a T38 fax session do not check RTP timeout */
if (dialog->t38.state == T38_ENABLED) {
dialog_unlink_rtpcheck(dialog);
return;
return CMP_MATCH;
}
/* If the call is not in UP state return for later check. */
if (dialog->owner->_state != AST_STATE_UP) {
return;
return 0;
}
/* Store these values locally to avoid multiple function calls */
@ -26038,8 +26049,7 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
/* If we have no timers set, return now */
if (!keepalive && !timeout && !hold_timeout) {
dialog_unlink_rtpcheck(dialog);
return;
return CMP_MATCH;
}
/* Check AUDIO RTP keepalives */
@ -26065,7 +26075,7 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
* Don't block, just try again later.
* If there was no owner, the call is dead already.
*/
return;
return 0;
}
ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
dialog->owner->name, (long) (t - dialog->lastrtprx));
@ -26082,10 +26092,12 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
ast_rtp_instance_set_timeout(dialog->vrtp, 0);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
}
dialog_unlink_rtpcheck(dialog); /* finally unlink the dialog from the checkrtp container */
/* finally unlink the dialog from dialogs_rtpcheck. */
return CMP_MATCH;
}
}
}
return 0;
}
/*! \brief The SIP monitoring thread
@ -26129,15 +26141,20 @@ static void *do_monitor(void *data)
/* Check for dialogs needing to be killed */
t = time(NULL);
/* Check for dialogs with rtp and rtptimeout
* All Dialogs which have rtp are in dialogs_rtpcheck container*/
ao2_t_callback(dialogs_rtpcheck, OBJ_NODATA | OBJ_MULTIPLE, dialog_checkrtp_cb, &t,
"callback to check rtptimeout and hangup calls if necessary");
/* Check for dialogs marked to be destroyed
* All Dialogs which need Destroy are in dialogs_needdestroy container*/
ao2_t_callback(dialogs_needdestroy, OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy, &t,
"callback to check rtptimeout and hangup calls if necessary");
/*
* Check dialogs with rtp and rtptimeout.
* All dialogs which have rtp are in dialogs_rtpcheck.
*/
ao2_t_callback(dialogs_rtpcheck, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE,
dialog_checkrtp_cb, &t,
"callback to check rtptimeout and hangup calls if necessary");
/*
* Check dialogs marked to be destroyed.
* All dialogs with needdestroy set are in dialogs_needdestroy.
*/
ao2_t_callback(dialogs_needdestroy, OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy,
NULL, "callback to check dialogs which need to be destroyed");
/* XXX TODO The scheduler usage in this module does not have sufficient
* synchronization being done between running the scheduler and places