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chan_pjsip: Fix deadlock when masquerading PJSIP channels.

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Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.

A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade.  With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer.  Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.

* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.

* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.

* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.

ASTERISK-24356 #close
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/4034/
........

Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2014-10-03 17:39:50 +00:00
parent b67094624d
commit 6a844be566
18 changed files with 253 additions and 49 deletions
Download Patch File Download Diff File Expand all files Collapse all files

65
include/asterisk/res_pjsip_session.h
View File

@@ -87,6 +87,9 @@ struct ast_sip_session_media {
*/
struct ast_sip_session_delayed_request;
/*! \brief Opaque struct controlling the suspension of the session's serializer. */
struct ast_sip_session_suspender;
/*!
* \brief A structure describing a SIP session
*
@@ -96,43 +99,45 @@ struct ast_sip_session_delayed_request;
* to use the term "SIP session" to refer to the INVITE dialog itself.
*/
struct ast_sip_session {
/* Dialplan extension where incoming call is destined */
/*! Dialplan extension where incoming call is destined */
char exten[AST_MAX_EXTENSION];
/* The endpoint with which Asterisk is communicating */
/*! The endpoint with which Asterisk is communicating */
struct ast_sip_endpoint *endpoint;
/* The contact associated with this session */
/*! The contact associated with this session */
struct ast_sip_contact *contact;
/* The PJSIP details of the session, which includes the dialog */
/*! The PJSIP details of the session, which includes the dialog */
struct pjsip_inv_session *inv_session;
/* The Asterisk channel associated with the session */
/*! The Asterisk channel associated with the session */
struct ast_channel *channel;
/* Registered session supplements */
/*! Registered session supplements */
AST_LIST_HEAD(, ast_sip_session_supplement) supplements;
/* Datastores added to the session by supplements to the session */
/*! Datastores added to the session by supplements to the session */
struct ao2_container *datastores;
/* Media streams */
/*! Media streams */
struct ao2_container *media;
/* Serializer for tasks relating to this SIP session */
/*! Serializer for tasks relating to this SIP session */
struct ast_taskprocessor *serializer;
/* Requests that could not be sent due to current inv_session state */
/*! Non-null if the session serializer is suspended or being suspended. */
struct ast_sip_session_suspender *suspended;
/*! Requests that could not be sent due to current inv_session state */
AST_LIST_HEAD_NOLOCK(, ast_sip_session_delayed_request) delayed_requests;
/* When we need to reschedule a reinvite, we use this structure to do it */
/*! When we need to reschedule a reinvite, we use this structure to do it */
pj_timer_entry rescheduled_reinvite;
/* Format capabilities pertaining to direct media */
/*! Format capabilities pertaining to direct media */
struct ast_format_cap *direct_media_cap;
/* When we need to forcefully end the session */
/*! When we need to forcefully end the session */
pj_timer_entry scheduled_termination;
/* Identity of endpoint this session deals with */
/*! Identity of endpoint this session deals with */
struct ast_party_id id;
/* Requested capabilities */
/*! Requested capabilities */
struct ast_format_cap *req_caps;
/* Optional DSP, used only for inband DTMF detection if configured */
/*! Optional DSP, used only for inband DTMF detection if configured */
struct ast_dsp *dsp;
/* Whether the termination of the session should be deferred */
/*! Whether the termination of the session should be deferred */
unsigned int defer_terminate:1;
/* Deferred incoming re-invite */
/*! Deferred incoming re-invite */
pjsip_rx_data *deferred_reinvite;
/* Current T.38 state */
/*! Current T.38 state */
enum ast_sip_session_t38state t38state;
};
@@ -387,6 +392,28 @@ struct ast_sip_channel_pvt *ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_
struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint,
struct ast_sip_contact *contact, pjsip_inv_session *inv);
/*!
* \brief Request and wait for the session serializer to be suspended.
* \since 12.7.0
*
* \param session Which session to suspend the serializer.
*
* \note No channel locks can be held while calling without risk of deadlock.
*
* \return Nothing
*/
void ast_sip_session_suspend(struct ast_sip_session *session);
/*!
* \brief Request the session serializer be unsuspended.
* \since 12.7.0
*
* \param session Which session to unsuspend the serializer.
*
* \return Nothing
*/
void ast_sip_session_unsuspend(struct ast_sip_session *session);
/*!
* \brief Create a new outgoing SIP session
*