| Age | Commit message (Collapse) | Author |
|---|
|
This change fixes a bug where the DTLS timeout timer would be
initialized to 0 if DTLS was not used for an RTP session.
ASTERISK-25103
Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac
|
|
mwi_subscription_mailboxes_str()." into 13
|
|
|
|
into 13
|
|
* changes:
res_pjsip_mwi.c: Eliminate a simple RAII_VAR.
res_pjsip_mwi.c: Fix mid-line log message line breaks.
|
|
dereferences." into 13
|
|
|
|
transaction." into 13
|
|
|
|
This change moves logic for setting up the DTLS SSL contexts to
when the SDP is done being processed instead of when ICE negotiation
completes. It also stops handshakes from being initiated when we
are acting as a server.
Manipulating the SSL context when ICE negotiation has completed
is problematic as the SSL context is not protected and if acting
as a client the remote side may have started DTLS negotiation
already.
The retransmission timeout timer code has also been split up
and simplified some. Both RTP and RTCP now have their own timers
and the points at which the timer is stopped and started is now
more specific. When a packet is sent the timer is started. When
a response is received but before it is processed the timer is
stopped. This provides a guarantee that the timeout is not
occurring while the response is processed.
ASTERISK-22805 #close
ASTERISK-24550 #close
ASTERISK-24651 #close
ASTERISK-24832 #close
ASTERISK-25103 #close
ASTERISK-25127 #close
Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91
|
|
Calling t38_change_state() sets the t38 state so it makes little sense to
then check the state right after the call for something else.
* Made the code in t38_interpret_parameters() reject or exit T.38 mode as
intended but not implemented.
Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2
|
|
Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907
|
|
Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da
|
|
* Add create_mwi_subscriptions_for_endpoint() doxygen comment.
Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2
|
|
MWI subscriptions can crash or corrupt memory when using the subscription
datastore to access the MWI subscription object because the datastore is
not holding a reference to the object.
* Give the subscription datastore a ref to the MWI subscription object.
It is unfortunate that the ref causes a circular ref chain that must be
explicitly broken to allow the memory to get released. The loop is broken
when the subscription is shutdown and if the subscription setup fails.
ASTERISK-25168 #close
Reported by: Carl Fortin
Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f
|
|
When res_pjsip body generator modules were generating XML or XPIDF
response bodies, there was a chance that the generated body would be the
exact size of the supplied buffer. Adding the nul string terminator would
then write beyond the end of the buffer and potentially corrupt memory.
* Fix MALLOC_DEBUG high fence violations caused by adding a nul string
terminator on the end of a buffer for XML or XPIDF response bodies.
* Made calls to pj_xml_print() safer if the XML prolog is requested. Due
to a bug in pjproject, the return value could be -1 _or_
AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.
* Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
return value of pj_xml_print() when the supplied buffer is not large
enough.
ASTERISK-25168
Reported by: Carl Fortin
Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de
|
|
When a caller calls a FAX number and then hangs up right after the call is
answered then the T.38 re-INVITE automatic reject timer may still be
running after the channel goes away.
* Added session NULL channel checks on the code paths that get executed by
t38_automatic_reject() to prevent a crash when the T.38 re-INVITE
automatic reject timer expires.
ASTERISK-25168
Reported by: Carl Fortin
Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403
|
|
All send/receive processing for a SIP transaction needs to be done under
the same threadpool serializer to prevent reentrancy problems inside
pjproject and res_pjsip.
* Add threadpool API call to get the current serializer associated with
the worker thread.
* Pick a serializer from a pool of default serializers if the caller of
res_pjsip.c:ast_sip_push_task() does not provide one.
This is a simple way to ensure that all outgoing SIP request messages are
processed under a serializer. Otherwise, any place where a pushed task is
done that would result in an outgoing out-of-dialog request would need to
be modified to supply a serializer. Serializers from the default
serializer pool are picked in a round robin sequence for simplicity.
A side effect is that the default serializer pool will limit the growth of
the thread pool from random tasks. This is not necessarily a bad thing.
* Made pjsip_distributor.c save the thread's serializer name on the
outgoing request tdata struct so the response can be processed under the
same serializer.
This is a cherry-pick from master.
**** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a
NOTE: session_inv_on_state_changed() is disassociating the dialog from the
session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED.
Unfortunately this is a tad too soon because our BYE request transaction
has not completed yet.
ASTERISK-25183 #close
Reported by: Matt Jordan
Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a
|
|
This change makes it so that when accepting a WebSocket
connection the HTTP response is sent as one packet instead of
fragmented. Browsers don't like it when you send it fragmented.
ASTERISK-25103
Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69
|
|
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602
|
|
This fixes so a failure to get a timer file descriptor does not cascade
to closing FD 0.
On error, both res_timing_kqueue and res_timing_timerfd would call the
destructor before setting the file handle. The file handle had been
initialized to 0, causing FD 0 to be closed. This in turn, resulted in
floods of "CLI>" messages and an unusable terminal.
ASTERISK-19277 #close
Reported by: Barry Chern
For the 13 branch, this was already fixed. This patch only ensures that
we do not attempt to close a negative file descriptor.
Change-Id: I147d7e33726c6e5a2751928d56561494f5800350
|
|
This prevents a leak of a sorcery object type when realtime sorcery
objects are retrieved by fields or when multiple objects are retrieved.
The extent of this leak is that sorcery object types would be leaked.
These are allocated whenever an object type is registered with sorcery,
meaning that on module shutdown, these objects would be leaked. This
could be problematic if many reloads were performed, but it is not as
severe as if every sorcery object retrieved from realtime were being
leaked.
ASTERISK-25165 #close
Reported by Corey Farrell
Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8
|
|
|
|
Returns a 'failure' from the module load routine indicates to Asterisk
that it should abort loading completely. This is rarely - in fact,
really, never - a good option. Aborting load of Asterisk from a dynamic
module implies that the core, and the rest of the dynamic modules, don't
matter: we should abandon all processing.
res_corosync is really not that important.
This patch updates the module such that, if it fails to load, it
politely declines (emitting ERROR messages along the way), and allows
Asterisk to continue to function.
Note that this issue was keeping Asterisk unit tests from running on
certain build agents.
Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f
|
|
A previous change made the contact only get rewritten if the dialog's
route set was not marked frozen. Unfortunately, while the intent of this
is correct, the dialog's route set actually gets marked as frozen
earlier than expected, especially for UAS dialogs.
Instead, the idea is that the contact needs to not be rewritten if there
is a pre-existing route set on the dialog. This is now accomplished by
checking the dialog's route set list instead of checking if the route
set is frozen.
Doing this causes some broken tests to begin passing again.
ASTERISK-25196
Reported by Mark Michelson
Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e
|
|
The client_state objects contain a serializer used to send the outbound
REGISTER messages. Once all those message transactions are complete then
the module can shutdown.
ASTERISK-24907 #close
Reported by: Kevin Harwell
Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547
|
|
|
|
handle_client_state_destruction() refs" into 13
|
|
ast_sorcery_object_unregister() API" into 13
|
|
|
|
unload_module()." into 13
|
|
res_pjsip_refer will attempt to add Referred-By or Replaces headers to
outbound INVITEs at times. If the INVITE gets challenged for
authentication, then we will resend the INVITE. Prior to this patch, the
Referred-By or Replaces header would be re-added to the outbound INVITE,
resulting in duplicated headers.
ASTERISK-25204 #close
Reported by Mark Michelson
Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d
|
|
When performing some provider testing, the rewrite_contact option was
interfering with proper construction of a route set when sending an ACK
after receiving a 200 OK response to an INVITE.
The initial INVITE was sent to address sip:foo. The 200 OK had a Contact
header with URI sip:bar. In addition, the 200 OK had Record-Route
headers for sip:baz and sip:foo, in that order. Since the Record-Route
headers had the lr parameter, the result should have been:
* Set R-URI of the ACK to sip:bar.
* Add Route headers for sip:foo and sip:baz, in that order.
However, the rewrite_contact option resulted in our rewriting the
Contact header on the 200 OK to sip:foo. The result was:
* R-URI remained sip:foo.
* We added Route headers for sip:foo and sip:baz, in that order.
The result was that sip:bar was not indicated in the ACK at all, so the
far end never received our ACK. The call eventually dropped.
The intention of rewrite_contact is to rewrite the most immediate
destination of our SIP request to be the same address on which we
received a request or response. In the case of processing a SIP response
with Record-Route headers, this means that instead of rewriting the
Contact header, we should instead rewrite the bottom-most Record-Route
header. In the case of processing a SIP request with Record-Route
headers, this means we rewrite the top-most Record-route header.
Like when we rewrite the Contact header, we also ensure to update
the dialog's route set if it exists.
ASTERISK-25196 #close
Reported by Mark Michelson
Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f
|
|
A module trying to unload needs to wait for all serializers it creates and
uses to complete processing before unloading.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059
|
|
* handle_client_state_destruction() must always be passed a ref to
client_state because it will always unref client_state.
handle_registration_response() was not passing a client_state ref.
* Made the final un-REGISTER message get sent normally using the pjproject
register control structure in handle_client_state_destruction(). The
previous code attempted to short circuit the response handling for the
module to unload. That doesn't work for a couple reasons. One,
pjsip_regc_send() may call the registered callback before it returns and
unbalance the client_state ref count. Two, the registered callback
handles any authentication for the un-REGISTER message.
* Made the distinction between internal registration state and external
registration status with sip_outbound_registration_status_str(). This is
necessary to avoid altering documented AMI messages with internal
changes.
* Removed references to client_state->client outside of the serializer
thread. When handle_client_state_destruction() destroys the pjproject
register control structure that memory is freed and cannot be referenced
anymore. These accesses were to provide information for debug and
off-nominal warning messages.
* In sip_outbound_registration_timer_cb() you should not access entry->id
after unrefing client_state because the passed in entry is normally
pointing to the timer entry in the client_state object.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
|
|
The sorcery pjsip 'registration' config object needs to be destroyed on
module unload. Otherwise, a reload of res_pjsip could try to use
callbacks for a previously unloaded instance of the module provided by
ast_sorcery_object_register() or one of the variants. Also, if
res_pjsip_outbound_registration were subsequently reloaded, the sorcery
config field objects would be registered in sorcery twice.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697
|
|
It is best if the loading code creates and initializes the module's
infrastructure before letting the system know of its existence. The
unloading code needs to reverse the actions of the loading code and in the
reverse order.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4
|
|
|
|
The res_pjsip_mwi previously required a reload to set up the proper
subscriptions to allow unsolicited MWI to work. This change
makes it so the act of registering will also cause this to occur.
This is particularly useful if realtime is involved as no reload
needs to occur within Asterisk to cause the MWI information
to get sent.
ASTERISK-25180 #close
Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252
|
|
Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7
|
|
Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e
|
|
* Break some long lines.
* Fix doxygen comment.
Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305
|
|
This resolves two observed race conditions.
First, a bit of background on what the Stasis application does:
1a Creates a stasis_app_control structure. This structure is linked into
a global container and can be looked up using a channel's unique ID.
2a Puts the channel in an event loop. The event loop can exit either
because the stasis_app_control structure has been marked done, or
because of some other factor, such as a hangup. In the event loop, the
stasis_app_control determines if any specific ARI commands need to be
run on the channel and will run them from this thread.
3a Checks if the channel is bridged. If the channel is bridged, then
ast_bridge_depart() is called since channels that are added to Stasis
bridges are always imparted as departable.
4a Unlink the stasis_app_control from the container.
When an ARI command is received by Asterisk, the following occurs
1b A thread is spawned to handle the HTTP request
2b The stasis_app_control(s) that corresponds to the channel(s) in the
request is/are retrieved. If the stasis_app_control cannot be
retrieved, then it is assumed that the channel in question has exited
the Stasis app or perhaps was never in Stasis in the first place.
3b A command is queued onto the stasis_app_control, and the channel's
event loop thread is signaled to run the command.
4b While most ARI commands do nothing further, some, such as adding or
removing channels from a bridge, will block until the command they
issued has been completed by the channel's event loop.
The first race condition that is solved by this patch involves a crash
that can occur due to faulty detection of the channel's bridged status
in step 3a. What can happen is that in step 2a, the event loop may run
the ast_bridge_impart() function to asynchronously place the channel
into a bridge, then immediately exit the event loop because the channel
has hung up. In step 3a, we would detect that the channel was not
bridged and would not call ast_bridge_depart(). The reason that the
channel did not appear to be bridged was that the depart_thread that is
spawned by ast_bridge_impart() had not yet started. That is the thread
where the channel is marked as being bridged. Since we did not call
ast_bridge_depart(), the Stasis application would exit, and then the
channel would be destroyed Then the depart_thread would start up and
try to manipulate the destroyed channel, causing a crash.
The fix for this is to switch from using ast_channel_is_bridged() to
checking the NULLity of ast_channel_internal_bridge_channel() to
determine if ast_bridge_depart() needs to be called. The channel's
internal bridge_channel is set when ast_bridge_impart() is called and
is NULLed by the call to ast_bridge_depart(). If the channel's internal
bridge_channel is non-NULL, then the channel must have been imparted
into the bridge and needs to be departed, even if the actual bridging
operation has not yet started. By departing the channel when necessary,
the thread that is running the Stasis application will block until the
bridge gives the okay that the depart_thread has exited.
The second race condition that is solved by this patch involves a leak
of HTTP handler threads. The problem was that step 2b would successfully
retrieve a stasis_app_control structure. Then step 2a would exit the
channel from the event loop due to a hangup. Steps 3a and 4a would
execute, and then finally steps 3b and 4b would. The problem is that at
step 4b, when attempting to add a channel to a bridge, the thread would
block forever since the channel would never execute the queued command
since it was finished with the event loop. This meant that the HTTP
handling thread would be leaked, along with any references that thread
may have owned (in my case, I was seeing bridges leaked).
The fix for this is to hone in better on when the channel has exited the
event loop. The stasis_app_control structure has an is_done field that
is now set at each point where the channel may exit the event loop. If
step 2b retrieves a valid stasis_app_control structure but the control
is marked as done, then the attempted operation exits immediately since
there will be nothing to service the attempted command.
ASTERISK-25091 #close
Reported by Ilya Trikoz
Change-Id: If66265b73b4c9f8f58599124d777fedc54576628
|
|
This event was added some time ago in order to clarify when a channel
took the place of another channel in a parking lot. However, there was
no XML documentation added for the event. This patch adds the XML
documentation.
ASTERISK-24900 #close
Reported by Rusty Newton
Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac
|
|
Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
|
|
This patch fixes use-after-free bugs caught by AddressSanitizer.
1. PJSIP transport manager may decide to destroy transport on its own.
For example, when the contact registered via websocket has not renewed
its registration in time. The transport was destoyed, but the websocket
listener thread was still active until the socket closes, and then tried
to call transport_shutdown on transport that has been freed.
Also, the transport destructor accessed wstransport->rdata.tp_info.pool
right after freeing memory that contained wstransport itself.
This patch converts transport to an ao2 object, allowing it to be
refcounted, so that it is available until both websocket listener and
pjsip transport manager are finished with it.
2. The websocket listener deletes the last reference on websocket session
when the tcp connection is closed, and it gets destroyed, but
the transport manager may still use it, for example when disconnect
happens in the middle of a SIP transaction.
A new reference to websocket session has been added that is released
with the transport to prevent this.
ASTERISK-25096 #close
Reported by: Josh Kitchens
ASTERISK-24963 #close
Reported by: Badalian Vyacheslav
Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b
|
|
|
|
It is possible to receive incoming requests or responses after the channel
on an ast_sip_session has been destroyed and NULLed out. Handlers of these
sorts of requests or responses need to be prepared for the possibility
that the channel is NULL or else they could cause a crash.
While several places have been amended to deal with NULL channels, there
were still a couple of places that needed updating.
res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to
return early if there is no channel on the session.
res_pjsip_session.c: When handling a 302 response, we need to stop the
redirecting attempt if there is no channel on the session.
ASTERISK-25148 #close
reported by Mark Michelson
Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9
|
|
contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status
to force the creation of a contact_status object whenever a new
contact is added but it didn't unref the returned object.
Added an ao2_cleanup(status) to plug the leak.
ASTERISK-25141
Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40
Reported-by: Corey Farrell
|
|
* Add some type casting so tv_usec can really be a long, instead of
some strange platform specific type.
* Add some .dylib style files to .gitignore.
* Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer
versions of GCC, when compiling the Homebrew formula for Asterisk,
are not properly passing the -Xlinker options to the linker. Given
that -Wl, does exactly the [same thing][], and does it properly, this
patch changes the -Xlinker options to use -Wl, instead.
[reasons unknown]: http://bit.ly/1SUbEYx
[same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html
Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd
|
