Beside that, the format-attribute module sends only non-default values in the
line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore,
previously the parameter stereo was not parsed when being the first parameter.
ASTERISK-25583 #close
Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73
When 90d9a70789 was merged, it mostly tested dynamic contacts created as
a result of registering a PJSIP endpoint. Contacts generated in this
fashion typically have a long alphanumeric string as their object identifier,
which maps reasonably well for StatsD. Unfortunately, this doesn't work in the
general case. StatsD treats both '.' and ':' characters as special characters.
In particular, having a ':' appear in the middle of a StatsD metric will
result in the metric being rejected.
This causes some obvious issues with SIP URIs.
The StatsD API should not be responsible for escaping the metric name passed
to it. The metric is treated as a single long string, and it would be
challenging to know what to escape in the string passed to the function.
Likewise, we don't want to escape the metric in PJSIP, as that involves
overhead that is wasted when either res_statsd isn't loaded or enabled.
This patch takes an alternative approach. The Contact ID has been changed
to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the
aforementioned special characters, (b) can be done on Contact creation,
which has minimal impact on run-time performance, and (c) also conforms to an
earlier commit that changed the ID for dynamic contacts.
The downside of this is that StatsD users will have to map SHA1 hashes back to
the Contacts that are emitting the statistics. To that end, the CLI commands
have been updated to include the first 10 characters of the MD5 hash, which
should be enough to match what is shown in Graphite (or some other StatsD
backend).
ASTERISK-25595 #close
Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2
Reported-by: Matt Jordan
Tested-by: George Joseph
An earlier commit changed the id of dynamic contacts to contain
a hash instead of the uri. This patch updates status change
logging to show the aor/uri instead of the id. This required
adding the aor id to contact and contact_status and adding
uri to contact_status. The aor id gets added to contact and
contact_status in their allocators and the uri gets added to
contact_status in pjsip_options when the contact_status is
created or updated.
ASTERISK-25598 #close
Reported-by: George Joseph
Tested-by: George Joseph
Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511
The fastagi record-file testsuite test sometimes fails reporting an empty
recorded file. This was happening because Asterisk was sending the agi result
notification prior to actually closing the file and the data, being buffered,
had not been written to the file yet when the test attempts to check the file
size.
This patch makes it so the record file stream is closed prior to sending the
agi result notification.
ASTERISK-25593 #close
Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde
The usage info for 'pjsip send notify' previously referenced the
chan_sip configuration sip_notify.conf. Fix this to reference
the correct configuration pjsip_notify.conf.
ASTERISK-25590 #close
Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea
This patch adds a module that emits StatsD statistics about Asterisk
endpoints. This includes:
* A GAUGE statistic for endpoint states, tracking how many endpoints are in
a particular state.
* A GAUGE statistic for each endpoint, counting the number of channels
currently associated with an endpoint.
ASTERISK-25572
Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305
If the sorcery object type is not found a NULL is returned.
Unfortunately, sorcery_realtime_filter_objectset() will crash after
complaining about not finding the object type and saying to expect errors.
* Use ao2_cleanup() instead of ao2_ref() to prevent the crash.
ASTERISK-25165
Reported by Corey Farrell
Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97
This patch adds the ability to send StatsD statistics related to the
state of PJSIP contacts. This includes:
* A GUAGE statistic measuring the count of contacts in a particular state.
This measures how many contacts are reachable, unreachable, etc.
* The RTT time for each contact, if those contacts are qualified. This
provides StatsD engines useful time-based data about each contact.
ASTERISK-25571
Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c
This patch adds outbound registration statistics for StatsD. This includes
the following:
* A GUAGE metric for the overall count of outbound registrations.
* A GUAGE metric for each state an outbound registration can be in. As the
outbound registrations change state, the overall count of how many
outbound registrations are in the particular state is changed.
These statistics are particularly useful for systems with a large number of
SIP trunks, and where measuring the change in state of the trunks is useful
for monitoring.
ASTERISK-25571
Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37
Often, the metric names of statistics we are generating for StatsD have some
dynamic component to them. This can be the name of a particular resource, or
some internal status label in Asterisk. With the current set of functions,
callers of the statsd API must first build the metric name themselves, then
pass this to the API functions. This results in a large amount of boilerplate
code and usage of either fixed length static buffers or dynamic memory
allocation, neither of which is desireable.
This patch adds two new functions to the StatsD API that support a printf
style format specifier for constructing the metric name. A dynamic string,
allocated in threadstorage, is used to build the metric name. This eases
the burden on users of the StatsD API.
Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea
When a channel is in a direct media bridge, a re-INVITE may arrive that forces
Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
must change its technology to a simple bridge, and re-INVITE the media back
to Asterisk.
Generally, this logic mostly already exists in Asterisk. However, prior to this
patch, there were a few bugs:
(1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
ever entering into a direct media bridge. This applies even when the only
media being passed over the channel is audio. This patch fixes this bug
by having the framehook specify that it defers caring about any frame type.
This allows the channels to enter into a direct media bridge, which will
be broken when a re-INVITE is received.
(2) When a re-INVITE is received, nothing instructed the bridging layer to
re-inspect the allowed bridging technology. This now occurs when either
a re-INVITE is received from a peer, or when a response is received from
the far end (that is, when the T.38 state changes to either
T38_PEER_REINVITE or T38_LOCAL_REINVITE).
(3) chan_pjsip needs to do a small amount of work to prevent a direct media
bridge from being chosen when a T.38 session is in progress. When a T.38
session supplement has a t38 datastore - which is added when we detect
we should start thinking about T.38 on a channel - we now refuse a native
RTP bridge.
(4) When a BYE request is received, we don't terminate the T.38 session. If
the other side of a T.38 fax survives the hangup (due to the 'g' flag
in Dial, for example), we don't currently re-INVITE the media on the
other channel back to audio. This patch now has res_pjsip_t38 intercept
BYE requests and inform the far side that the T.38 session is terminated.
This naturally causes the correct re-INVITEs to be sent.
ASTERISK-25582
Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb
This patch adds some debug statements to res_pjsip_t38. These statements help
to determine which SDP negotiation callbacks are being executed, and, when
a particular callback exits, why a callback may not have applied its logic
to the local or remote SDP.
Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77
When Asterisk is configured to use a dynamic sorcery backend (such as
res_sorcery_astdb) with 'registration' objects, it will fail to create the
internal state objects associated with the registration objects on module
load. This is due to nothing actually querying for the specific objects
and calling their sorcery apply handler during module load.
This patch fixes that by calling get_registrations in the sorcery observer's
object_type_loaded handler. Doing this causes the sorcery backends to be
asked for the current state of all registration objects, which causes the
apply handler to be called and the internal run-time state to be created.
ASTERISK-25575 #close
Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23
When no parameter is present, Asterisk does not generate the line fmtp, as
expected. However, because a buffer was reset, even rtpmap and fmtp of previous
media codecs got removed. Now, Asterisk does not reset other codecs in case of
no parameter for H.264.
ASTERISK-25573 #close
Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286
Receiving a 423 Interval Too Brief response after authentication for an
outbound registration attempt results in assuming that the registrar has
rejected the registration permanently. If there are no configured retries
for fatal responses then the outbound registration is stopped for that
endpoint.
For registrations, PJSIP/PJPROJECT intercepts the handling of 423
responses and does not include any authentication in the updated
registration request. When the updated request is challenged then the
Asterisk code assumes that we were challenged again because the peer
rejected the authentication we sent earlier.
* Made registration challenges keep track of the CSeq number to determine
if the received challenge response was for the request we thought we sent.
If the response's CSeq number differs from the CSeq number we last sent
with authentication then authenticate again because it is a challenge to a
different request.
Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09
When a request is sent using pjsip_endpt_send_request and fails, a condition
exists where the request wrapper, which is an AO2 object, may be de-ref'd
more times than it should. This occurs when the request's callback is called,
and, in the callback, the timer on the PJSIP heap is cancelled. When that
occurs, the request wrapper's lifetime is decremented. When
pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of
the request wrapper again, even though we've already cancelled the reference
associated with the timer.
This patch checks the return result of pj_timer_heap_cancel_if_active before
removing the reference associated with the timer. We now only decrement it
in this case if a timer is cancelled as a result of the function call.
Change-Id: I21332343a1a019c1117076f9bf2df27be2850102
If an authenticated incoming caller does not respond to our 200 OK INVITE
response with an ACK then PJSIP will hangup the call. Unfortunately,
there is a chance that the session's channel will go away between one use
of the channel pointer and another when building the BYE request because
the BYE is being built by the monitor thread and not the call's serializer
thread.
* Added a check to ensure that the thread trying to add the Reason header
is the call's serializer thread. This ensures that the channel will not
go away on us.
Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89
In practical tests, we have seen certain taskprocessors, specifically
Stasis subscription taskprocessors, cross the recently-added high-water
mark and emit a warning. This high-water mark warning is only intended
to be emitted when things have tanked on the system and things are
heading south quickly. In the practical tests, the Stasis taskprocessors
sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in
any danger at all.
As such, this ups the high-water mark to 500 tasks instead. It also
redefines the SIP threadpool request denial number to be a multiple of
the taskprocessor high-water mark.
Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce
When the SIP threadpool is backed up with tasks, we send 503 responses
to ensure that we don't try to overload ourselves. The problem is that
we were not insuring that we were not trying to send a 503 to an
incoming SIP response.
This change makes it so that we only send the 503 on incoming requests.
Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404
We have observed situations where the SIP threadpool may become
deadlocked. However, because incoming traffic is still arriving, the SIP
threadpool's queue can continue to grow, eventually running the system
out of memory.
This change makes it so that incoming traffic gets rejected with a 503
response if the queue is backed up too much.
Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816
When ASTERISK-25449 was closed, a number of scheduler issues mentioned in
the comments were missed. These have since beed raised in ASTERISK-25476
and elsewhere.
This patch attempts to collect all of the scheduler issues discovered so
far and address them sensibly.
ASTERISK-25476 #close
Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b
In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual
amount of channels is negotiated in-band. Therefore now, the Opus codec and its
attribute rtpmap are registered with two channels.
ASTERISK-24779 #close
Reported by: PowerPBX
Tested by: Alexander Traud
patches:
asterisk-24779.patch submitted by Sean Bright (license #5060)
Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b
Added a new api to res_statsd.c to allow it to receive a
character pointer for the value argument. This allows for a
'+' and a '-' to easily be sent with the value.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611
When an endpoint is deleted (such as through an API), the persistent endpoint
currently continues to lurk around. While this isn't harmful from a memory
consumption perspective - as all persistent endpoints are reclaimed on
shutdown - it does cause Stasis endpoint related operations to continue
to believe that the endpoint may or may not exist.
This patch causes the persistent endpoint related to a PJSIP endpoint to be
destroyed if the PJSIP endpoint is deleted.
Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb
The contact_status Sorcery objects are currently not destroyed when a contact
is deleted. This causes the contact's last known RTT/status to be 'sticky'
when the contact itself may no longer exist. This patch causes the
contact_status objects associated with both dynamic and static contacts to
be destroyed if the AoR holding those contacts is also destroyed (or via
other paths where a contact may be deleted.)
Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e
During a stress test of subscriptions, a huge blast of
subscription-related traffic resulted in the threadpool expanding to a
ridiculous number of threads. The balooning of threads resulted in an
increase of memory, which led to a crash due to being out of memory.
An easy fix for the particular test was to limit the size of the
threadpool, thus reining in the amount of memory that would be used. It
was decided that there really is no downside to having a non-infinite
default value for the maximum size of the threadpool, so this change
introduces 50 threads as the maximum threadpool size for the SIP
threadpool.
ASTERISK-25513 #close
Reported by John Bigelow
Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be
When an AoR is created or destroyed dynamically, the scheduled OPTIONS
requests that qualify the contacts on the AoR are not necessarily started
or destroyed, particularly for persistent contacts created for that AoR.
This patch adds create/update/delete sorcery observers for an AoR, which
schedule/unschedule the qualifies as expected.
Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d
When compiled with assertions enabled one will occur when destroying
the subscription tree when UAS dialog creation fails. This is because
the code assumes that a dialog will always exist on a subscription
tree when in reality during this specific scenario it won't.
This change makes it so a dialog is not removed from the subscription
tree if it is not present.
ASTERISK-25505 #close
Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee
Add the ability to filter output from pjsip list and show commands
using the "like" predicate like chan_sip.
For endpoints, aors, auths, registrations, identifyies and transports,
the modification was a simple change of an ast_sorcery_retrieve_by_fields
call to ast_sorcery_retrieve_by_regex. For channels and contacts a
little more work had to be done because neither of those objects are
true sorcery objects. That was just removing the non-matching object
from the final container. Of course, a little extra plumbing in the
common pjsip_cli code was needed to parse the "like" and pass the regex
to the get_container callbacks.
Some of the get_container code in res_pjsip_endpoint_identifier was also
refactored for simplicity.
ASTERISK-25477 #close
Reported by: Bryant Zimmerman
Tested by: George Joseph
Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1
During outbound registration it is possible to receive a fatal (any permanent/
non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due
to a problem with the registrar itself. Upon receiving the failure response
Asterisk terminates outbound registration for the given endpoint.
This patch adds an option, 'fatal_retry_interval', that when set continues
outbound registration at the given interval up to 'max_retries' upon receiving
a fatal response.
ASTERISK-25485 #close
Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2
A certain situation can result in our attempting to send a NOTIFY on a
destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but
that subscriber has dropped off the network. We end up retransmitting
that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY
transaction. When the pjsip evsub code is told that the transaction has
been terminated, it responds in kind by alerting us that the
subscription has been terminated, destroying the subscription, and then
removing its reference to the dialog, thus destroying the dialog.
The problem is that when we get told that the subscription is being
terminated, we detect that we have not sent a terminating NOTIFY
request, so we queue up such a NOTIFY to be sent out. By the time that
queued NOTIFY gets sent, the dialog has been destroyed, so attempting to
send that NOTIFY can result in a crash.
The fix being introduced here is actually a reintroduction of something
the pubsub code used to employ. We hold a reference to the dialog and
wait to decrement our reference to the dialog until our subscription
tree object is destroyed. This way, we can send messages on the dialog
even if the PJSIP evsub code wants to terminate earlier than we would
like.
In doing this, some NULL checks for subscription tree dialogs have been
removed since NULL dialogs are no longer actually possible.
Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5
When sending a NOTIFY, we lock the dialog and then unlock the dialog
when finished. A recent change made it so that the subscription tree's
dialog pointer will be set NULL when sending the final NOTIFY request
out. This means that when we attempt to unlock the dialog, we pass a
NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog
remains locked after we think we have unlocked it. When a response to
the NOTIFY arrives, the monitor thread attempts to lock the dialog, but
it cannot because we never released the dialog lock. This results in
Asterisk being unable to process incoming SIP traffic any longer.
The fix in this patch is to use a local pointer to save off the pointer
value of the subscription tree's dialog when locking and unlocking the
dialog. This way, if the subscription tree's dialog pointer is NULLed
out, the local pointer will still have point to the proper place and the
dialog lock will be unlocked as we expect.
Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a
The SIP dialog is removed from the subscription tree when the final
NOTIFY is sent. However, after the final NOTIFY is sent, the persistence
update function still attempts to access the cseq from the dialog,
resulting in a crash.
This fix removes the subscription persistence at the same time that the
dialog is removed from the subscription tree. This way, there is no
attempt to update persistence when the subscription is being destroyed.
Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb
There have been crashes seen where a taskprocessor's listener is NULL
unexpectedly.
Looking at backtraces, the problem was specifically seen in PJSIP
serializers.
Subscriptions make the mistake of removing a serializer from a dialog
during subscription tree destruction. Since subscription trees are
reference-counted, guaranteeing the circumstances behind the destruction
are not possible. This makes it so that the dialog serializer can be
removed while not holding the dialog lock. This makes it possible for
the distributor to get a pointer to the dialog serializer and have that
serializer get freed out from under it.
The fix for this is to remove the serializer from a subscription dialog
when sending the final NOTIFY. This guarantees that the serializer is
removed with the dialog lock held. By doing this, we guarantee that if
the distributor gains access to the dialog's serializer, it will not be
possible for the serializer to get freed by another thread.
Change-Id: I21f5dac33529f65cec45679bdace60670800ff66
If an old persistent subscription is recreated but then immediately
destroyed because it is out of date, the subscription tree will have no
leaf subscriptions on it. This was resulting in a crash when attempting
to destroy the subscription tree.
A simple NULL check fixes this problem.
Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac
There have been crashes and general instability seen in the pubsub code,
so this patch introduces three changes to increase the stability.
First, the ownership model for subscriptions has been modified. Due to
RLS, subscriptions are stored in memory as a tree structure. Prior to my
patch, the PJSIP subscription was the owner of the subscription tree.
When the PJSIP subscription told us that it was terminating, we started
destroying the subscription tree along with all of the individual leaf
subscriptions that belong to the tree. The problem with this model is
that the two actors in play here, the PJSIP subscription and the
individual leaf subscriptions, need to have joint ownership of the
subscription tree. So now, the PJSIP subscription and the individual
leaf subscriptions each have a reference to the subscription tree. This
way, we will not actually free memory until no players are left that
care. The PJSIP subscription is a bigger stakeholder, in that if the
PJSIP subscription's reference to the subscription tree is removed, the
subscription tree instructs the leaf subscriptions to shut down and drop
their references to the subscription tree when possible. The individual
leaf subscriptions, upon being told to shut down, can drop their stasis
subscriptions or whatever they use to learn of new state, and then drop
their reference to the subscription tree once they are ready to die.
Second, the lifetime of a PJSIP subscription's reference to our
subscription tree has been altered. As I learned from doing a deep dive,
the PJSIP evsub code can tell Asterisk multiple times that the
subscription has been terminated, and not all of these times
are especially helpful. I have altered the message flow that we use for
SIP subscriptions such that we will always drop the PJSIP subscription's
reference to the subscription tree when we send the NOTIFY that
terminates a SIP subscription. This also means that we will now queue
NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so
that we can have predictable state changes from the PJSIP evsub code.
Third, the synchronization of operations has been improved. PJSIP can
call into our code from a serializer thread (e.g. upon receiving an
incoming request) or from the monitor thread (e.g. when a subscription
times out). Because of this, there is the possibility of competing
threads stepping on each other. PJSIP attempts to do some
synchronization on its own by always keeping the dialog lock held when
it calls into us. However, since we end up pushing tasks into the
serializer, the result was that serialized operations were not grabbing
the dialog lock and could, as a result, step on something that was being
attempted by a different thread. Now we ensure that serialized
operations grab the dialog lock, then check for extenuating
circumstances, then proceed with their operation if they can.
Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5
In a realtime based system with a limited number of threadpool threads
it is possible for a deadlock to occur. This happens when permanent
endpoint state is updated, which will cause database queries to be done.
These queries may result in URI validation being done which is done
synchronously using a PJSIP thread. If all PJSIP threads are in use
processing traffic they themselves may be blocked waiting to get the
permanent endpoint container lock when identifying an endpoint.
This change moves URI validation to occur at use time instead of
configuration time. While this comes at a cost of not seeing a problem
until you use it it does solve the underlying deadlock problem.
ASTERISK-25486 #close
Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a
On v13, loading several thousand PJSIP endpoints on Asterisk start causes
a deadlock most of the time.
Thanks to mdu113 for discovering that there was a call to pgsql_exec() not
protected by the pgsql_lock reentrancy lock.
{quote}
I believe a code path exists that attempts to use pgsql connection without
locking pgsql_lock. I believe what happens during that deadlock that I
see is two concurrent threads are both attempting to send query to pgsql,
one of the thread is using a code path without locking pgsql_lock. If
they managed to send queries at the same time, it seems postgres ignores
one of the queries and replies only to the one of them. If it happens so
that the thread holding the lock didn't receive the reply it will wait for
it (and hold the lock) forever (or at least for very long time), thus
completely blocking all access to db.
{quote}
* Added missing reentrancy locking around pgsql_exec() in find_table().
* Moved unlock of pgsql_lock in unload_module() to avoid locking inversion
between the psql_tables list lock and the pgsql_lock.
ASTERISK-25455 #close
Reported by: mdu113
Patches:
res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113
Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2
The struct send_request_wrapper has a pjsip lock associated with it that
is created non-recursive. There is a code path for the struct
send_request_wrapper lock that will attempt to lock it recursively. The
reporter's deadlock showed that the thread calling endpt_send_request()
deadlocked itself right after the wrapper object got created.
Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited
MWI NOTIFY messages can hit this deadlock.
* Replaced the struct send_request_wrapper pjsip lock with the mutex lock
that can come with an ao2 object since all of Asterisk's mutexes are
recursive. Benefits include removal of code maintaining the pjsip
non-recursive lock since ao2 objects already know how to maintain their
own lock and the lock will show up in the CLI "core show locks" output.
ASTERISK-25435 #close
Reported by: Dmitriy Serov
Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d
In ast_rtp_read, the value of the variable 'mark' which we try to assign to a
frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate
it to 0 or 1.
ASTERISK-25451 #close
Change-Id: I53bdf5c026041730184a6a809009c028549ce626
When we decide we will no longer schedule an RTCP write, we remove the
reference to the RTP instance, then assign -1 to the stored scheduler ID
in case something else comes along and wants to see if anything is scheduled.
That scheduler ID is on the RTP instance. After 60a9172d7e was merged to
fix the regression introduced by 3cf0f29310, this improper assignment on a
potentially destroyed object started getting tripped on the build agents.
Frankly, this should have been crashing a lot more often earlier. I can only
assume that the timing was changed just enough by both changes to start
actually hitting this problem.
As it is, simply moving the assignment prior to the ao2 deference is sufficient
to keep the RTP instance from being referenced when it is very, truly,
aboslutely dead.
(Note that it is still good practice to assign -1 to the scheduler ID when we
know we won't be scheduling it again, as the ao2 deref *may* not always destroy
the ao2 object.)
ASTERISK-25449
Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7
Apparently some endpoints attempt to send a reINVITE before completing the
initial INVITE transaction. In this case PJSIP responds appropriately to
the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip
is using the initial INVITE transaction state to determine if an INVITE is
the initial INVITE or a reINVITE. Since the initial INVITE transaction
has not been confirmed yet chan_pjsip thinks the reINVITE is an initial
INVITE and starts another PBX thread on the channel. The extra PBX thread
ensures that hilarity ensues.
* Fix checks for a reINVITE on incoming requests to look for the presence
of a to-tag instead of the initial INVITE transaction state.
* Made caller_id_incoming_request() determine what to do if there is a
channel on the session or not. After a channel is created it is too late
to just store the new party id on the session because the session's party
id has already been copied to the channel's caller id.
ASTERISK-25404 #close
Reported by: Chet Stevens
Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be
When 5c713fdf18 was merged, it allowed for scheduled items to have an ID of
'0' returned. While this was valid per the documentation for the API, it was
apparently never returned previously. As a result, several users of the
scheduler API viewed the result as being invalid, causing them to reschedule
already scheduled items or otherwise fail in interesting ways.
This patch corrects the users such that they view '0' as valid, and a returned
ID of -1 as being invalid.
Note that the failing HEP RTCP tests now pass with this patch. These tests
failed due to a duplicate scheduling of the RTCP transmissions.
ASTERISK-25449 #close
Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39
A deadlock can happen when a sorcery object is being expired from the
memory cache when at the same time another object is being placed into the
memory cache. There are a couple other variations on this theme that
could cause the deadlock. Basically if an object is being expired from
the sorcery memory cache at the same time as another thread tries to
update the next object expiration timer the deadlock can happen.
* Add a deadlock avoidance loop in expire_objects_from_cache() to check if
someone is trying to remove the scheduler callback from the scheduler.
ASTERISK-25441 #close
Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc
Make sorcery_memory_cache_close() call remove_all_from_cache() instead of
partially inlining it.
ASTERISK-25441
Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c
Basically you should shutdown in the opposite order of how you setup since
later setup pieces likely depend on earlier setup pieces. e.g.,
Registering your external API with the rest of the system should be the
last thing setup and the first thing unregistered during shutdown.
Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e
In practice the set_role API callback can be invoked even
when no ICE is present on an RTP instance. This can occur
if ICE has not been enabled on it.
ASTERISK-25438 #close
Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69
* Now conf_alloc() has more off nominal error checking.
* Eliminated RAII_VAR() use in conf_alloc().
* Eliminated a dubius shortcut when destroying cfg->general in
conf_destructor() that would cause a crash if cfg->general failed to get
allocated.
* Add some ACO registration section comments.
Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a
Need to finish initializing the string fields in the ao2 object before
putting any default strings into them.
ASTERISK-25383 #close
Reported by: yaron nahum
Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84
When b99a705262 was merged, subscribing to a
NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to
all bridges. Unfortunately, the res_stasis control loop did not check that
a bridge changing on a channel's control object was actually also non-NULL.
As a result, app_subscribe_bridge will be called with a NULL bridge when a
channel leaves a bridge. This causes a new subscription to be made to the
bridge. If an application has also subscribed to the bridge, the application
will now have two subscriptions:
(1) The explicit one created by the app
(2) The implicit one accidentally created by the control structure
As a result, the 'BridgeDestroyed' event can be sent multiple times. This
patch corrects the control loop such that it only subscribes an application
to a new bridge if the bridge pointer is non-NULL.
ASTERISK-24870
Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f
This patch adds the ability to subscribe to all events. There are two possible
ways to accomplish this:
(1) On initial WebSocket connection. This patch adds a new query parameter,
'subscribeAll'. If present and True, Asterisk will subscribe the
applications to all ARI events.
(2) Via the applications resource. When subscribing in this manner, an ARI
client should merely specify a blank resource name, i.e., 'channels:'
instead of 'channels:12354'. This will subscribe the application to all
resources of the 'channels' type.
ASTERISK-24870 #close
Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6
This patch adds support for subscribing to all device state changes. This is
done either by subscribing to an empty device, e.g., 'eventSource=deviceState:',
or by the WebSocket connection specifying that it wants all state in the
system.
ASTERISK-24870
Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32
This patch adds support for receiving events regarding Peer status changes
and Contact status changes. This is particularly useful in scenarios where
we are subscribed to all endpoints and channels, where we often want to know
more about the state of channel technology specific items than a single
endpoint's state.
ASTERISK-24870
Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9
There is a slim chance of a race condition occurring where two threads
can both attempt to manipulate the same area.
Thread A can be handling an incoming initial SUBSCRIBE request. Thread A
lets the specific subscription handler know that the subscription has
been established.
At this point, Thread B may detect a state change on the subscribed
resource and queue up a notification task on Thread C, the subscription
serializer thread.
Now Thread A attempts to generate the initial NOTIFY request to send to
the subscriber at the same time that Thread C attempts to generate a
state change NOTIFY request to send to the subscriber.
The result is that Threads A and C can step on the same memory area,
resulting in a crash. The crash has been observed as happening when
attempting to allocate more space to hold the body for the NOTIFY.
The solution presented here is to queue the subscription establishment
and initial NOTIFY generation onto the subscription serializer thread
(Thread C in the above scenario). This way, there is no way that a state
change notification can occur before the initial NOTIFY is sent, and if
there is a quick succession of NOTIFYs, we can guarantee that the two
NOTIFY requests will be sent in succession.
Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815
We should not try to send a SIP response message because we may be
restoring a persistent subscription where we are not responding to a SIP
request.
Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec
ast_sip_pubsub_register_body_generator() did not account for the null
terminator set by sprintf() in the allocated output buffer.
Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2
The default_from_user retrieval function was pulling the
default_from_user from the global configuration struct in an unsafe way.
If using a database as a backend configuration store, the global
configuration struct is short-lived, so grabbing a pointer from it
results in referencing freed memory.
The fix here is to copy the default_from_user value out of the global
configuration struct.
Thanks go to John Hardin for discovering this problem and proposing the
patch on which this fix is based.
ASTERISK-25390 #close
Reported by Mark Michelson
Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c
We will only rewrite the Contact header if there is no Record-Route header in
the received request. If a malfunctioning proxy places a Record-Route header
into a REGISTER request, we will decide that we shouldn't update the IP/port
in the Contact header, and we will end up storing a contact with an AoR that
contains the NAT'd IP address.
While it is nice to have the proxy *not* send a Record-Route in a REGISTER
request, it's also a good idea to not process the header in a non-dialog
message. This patch updates the code to explicitly ignore the Record-Route
header in REGISTER requests.
ASTERISK-25387 #close
Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f
Make certain that the pjsip session has not failed to
allocate the format capabilities structure, which can
otherwise cause a crash when referenced.
ASTERISK-25323
Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750
In working through a recent ICE negotiation bug, I found the debug
logging in res_rtp_asterisk to be lacking. This patch adds a number of
debug and warning statements that were helpful.
Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80
In the wild it is possible for Contact URIs to be quite long as
parameters can exist on them. This can present a problem when storing
them in the AstDB as the URI is used as part of the object name and
there is a fixed length limit for the AstDB. This will cause
the contact to not get stored.
This change uses the MD5 hash of the Contact URI as part of the
object name instead. This has a fixed length which is guaranteed
to not exceed the AstDB length limit.
ASTERISK-25295 #close
Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02
When an AoR is deleted by an external mechanism, such as through ARI, we
currently do not remove dynamic contacts that were created for that AoR as a
result of a received REGISTER request. As a result, re-creating the AoR will
cause the dynamic contact to be interpreted as a persistent contact, leading
to some rather strange state being created for the contacts/endpoints.
This patch adds a sorcery observer for the 'aor' object. When a delete is
issued on the underlying sorcery object, the observer is called, and all
contacts created and persisted in sorcery for that AoR are also removed. Note
that we don't want to perform this action when an AO2 object that is an AoR is
destroyed, as the AoR can still exist in the backing storage (and we would
thus be removing valid contacts from an AoR that still "exists".)
ASTERISK-25381 #close
Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328
We were passing the wrong count into pj_ice_sess_create_check_list(),
causing the create to fail if we ever received more than PJ_ICE_MAX_CAND
candidates.
Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378
When Asterisk sends an outbound SIP request, if there is no direct
reason to place a specific value for the username in the From header,
Asterisk would generate a UUID. For example, this would happen when
sending outbound OPTIONS requests when qualifying or when sending
outbound INVITE requests when originating (if no explicit caller ID were
provided). The issue is that some SIP providers reject these sorts of
requests with a "Name too long" error response.
This patch aims to fix this by changing the default outbound username in
From headers to "asterisk". This value can be overridden by changing the
default_from_user option in the global options if desired.
ASTERISK-25377 #close
Reported by Mark Michelson
Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190
In Asterisk 11, the announcer channel would receive channel variables
from the channel being parked by means of normal channel inheritance.
This functionality was lost during the big res_parking project in
Asterisk 12. This patch restores that functionality.
ASTERISK-25369 #close
Review: https://gerrit.asterisk.org/#/c/1180/
Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e
Pjsip is refusing to use unsecure transport with "sips" in url.
WSS should be considered as secure transport.
ASTERISK-24602 #comment Partially fixed by setting WSS as secure
Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353
When sending a stateful response, creation of the transaction can fail,
most commonly because we are trying to create a transaction from a
retransmitted request. When creation of the transaction fails, we end up
leaking a reference to a contact that was bumped when the response was
created.
This patch adds the missing deref and fixes the reference leak.
Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07
A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as
a means of writing an appropriate packet to persistent storage. While
this partially solved the issue, it had its own problems.
pjsip_msg_print will always add a Content-Length header to the message
it prints. Frequent restarts of Asterisk can result in persistent
subscriptions being written with five or more Content-Length headers. In
addition, sometimes some apparent corruption of individual headers could
be seen.
This aims to fix the problem by not running a parsed message through an
interpreter but rather by taking the raw message and saving it. The
logic for what to save is going to be different depending on whether a
SUBSCRIBE was received from the wire or if it was pulled from
persistence. When receiving a packet from the wire, when using a
streaming transport, the rdata->pkt_info.packet may contain multiple SIP
messages or fragments. However, the rdata->msg_info.msg_buf will always
contain the current SIP message to be processed. When pulling from
persistence, though, the rdata->msg_info.msg_buf will be NULL since no
transport actually handled the packet. However, since we know that we
will always ever pull one SIP message from persistence, we are free to
save directly from rdata->pkt_info.packet instead.
ASTERISK-25365 #close
Reported by Mark Michelson
Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b
The keepalive support in res_pjsip_sdp_rtp currently assumes
that a stream will only be negotiated once. This is false.
If the stream is replaced and later added back it can be
negotiated again causing multiple keepalive scheduled items
to exist. This change explicitly deletes the existing
keepalive scheduled item before adding the new one.
The res_pjsip_sdp_rtp module also does not stop RTP
keepalives or timeout timer if the stream has been
replaced. This change adds a callback to the session media
interface to allow a media stream to be stopped without
the resources being destroyed. This allows the scheduled
items and RTP to be stopped when the stream no longer
exists.
ASTERISK-25356 #close
Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de
When a BYE request is received the PJSIP invite session implementation
creates and sends a 200 OK response before we are aware of it. This
causes the INVITE session state callback to be called into and ultimately
the session supplements run on the BYE request. Once this response has
been sent the normal transaction state callback is invoked which
invokes the session supplements on the BYE request again. This can
be problematic in particular with res_pjsip_rfc3326 as it may
attempt to update the hangup cause code on the channel while it is
in the process of being hung up.
This change makes it so the session supplements are only invoked
once by the INVITE session state callback.
ASTERISK-25318 #close
Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a
If the ast_strndup() call fails to allocate a copy of the
transport string for parsing, fail gracefully.
ASTERISK-25323
Reported by: Scott Griepentrog
Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28
Modules commonly used the pj_gethostip function for retrieving the
IP address of the host. This function does not cache the result and may
result in a DNS lookup occurring, or additional work. If the DNS
server is unreachable or network issues arise this can cause the
pj_gethostip function to block for a period of time.
This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
function which does the same thing but caches the host IP address at
module load time. This results in no additional work being done each
time the local host IP address is needed.
ASTERISK-25342 #close
Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e
When recreating a subscription it is possible for a freed sub_tree
to be referenced when the initial NOTIFY fails to be created.
Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788
When an endpoint is backed by a non-static conf file backend (such as
the AstDB or Realtime), the 'auth' object may be returned as being an
empty string. Currently, res_pjsip will interpret that as being a valid
auth object, and will attempt to authenticate inbound requests. This
isn't desired; is an auth value is empty (which the name of an auth
object cannot be), we should instead interpret that as being an invalid
auth object and skip it.
ASTERISK-25339 #close
Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7
There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk. It uses only one mapping structure to
associate payload types to codecs. The single mapping is overkill if all
of the payload type values are well known values. Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive. Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.
1) An independent payload type mapping is needed for sending and
receiving.
2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.
3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.
* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.
* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.
* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created. All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.
* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP. We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
This is a type mismatch fix of the debugging commit
c63316eec1 made to find out why
a testsuite test was failing only on one of the continuous
integration build agents.
Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75
When sending an RTP keepalive, we need to be sure we're not dealing with
a NULL RTP instance. There had been a NULL check, but the commit that
added the rtp_timeout and rtp_hold_timeout options removed the NULL
check.
Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64
Due to the use of ast_websocket_close in session termination it is
possible for the underlying socket to already be closed when the
session is terminated. This occurs when the close frame is attempted
to be written out but fails.
Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b
The res_http_websocket module will currently attempt to close
the WebSocket connection if fatal cases occur, such as when
attempting to write out data and being unable to. When the
fatal cases occur the code attempts to write a WebSocket close
frame out to have the remote side close the connection. If
writing this fails then the connection is not terminated.
This change forcefully terminates the connection if the
WebSocket is to be closed but is unable to send the close frame.
ASTERISK-25312 #close
Change-Id: I10973086671cc192a76424060d9ec8e688602845
This patch adds the .get callback to the format attribute module, such
that the Asterisk core or other third party modules can query for the
negotiated format attributes.
Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c
If the saved SUBSCRIBE message is not parseable for whatever reason then
Asterisk could crash when libpjsip tries to parse the message and adds an
error message to the parse error list.
* Made ast_sip_create_rdata() initialize the parse error rdata list. The
list is checked after parsing to see that it remains empty for the
function to return successful.
ASTERISK-25306
Reported by Mark Michelson
Change-Id: Ie0677f69f707503b1a37df18723bd59418085256
We don't have a compatability function to fill in a missing htobe64; but
we already have one for the identical htonll.
Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac
An http request can be sent to get the existing Asterisk logs.
The command "curl -v -u user:pass -X GET 'http://localhost:8088
/ari/asterisk/logging'" can be run in the terminal to access the
newly implemented functionality.
* Retrieve all existing log channels
ASTERISK-25252
Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808
An http request can be sent to create a log channel
in Asterisk.
The command "curl -v -u user:pass -X POST
'http://localhost:088/ari/asterisk/logging/mylog?
configuration=notice,warning'" can be run in the terminal
to access the newly implemented functionality for ARI.
* Ability to create log channels using ARI
ASTERISK-25252
Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782
An http request can be sent to delete a log channel
in Asterisk.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/logging/mylog'" can be run in the terminal
to access the newly implemented functionally for ARI.
* Able to delete log channels using ARI
ASTERISK-25252
Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6
Matt Jordan
Alexander Traud
Joshua Colp
George Joseph
Richard Mudgett
Kevin Harwell
David M. Lee
Corey Farrell
Mark Michelson
Steve Davies
tcambron
StefanEng86
Scott Griepentrog
Jonathan Rose
Martin Tomec
Scott Emidy