bridge_native_rtp can call into an RTP-capable channel driver in order
for the driver to update information about who the channel is
communicating with. For SIP channel drivers, this means deactivating
RTCP and sending a reinvite so that the endpoints can communicate
directly.
bridge_native_rtp does the right thing and has the channel locked when
calling into the channel driver. chan_pjsip can't alter session
properties in this thread, though. chan_pjsip queues a task on the
session serializer in order to update properties there.
The problem is that this queued task was not locking the channel. This
meant that the queued task could attempt to deactivate RTCP at the same
time that the channel thread was attempting to process an incoming RTCP
packet. This could lead to a crash.
This patch fixes the issue by locking the channel in the queued task
when altering RTP properties.
ASTERISK-26092 #close
Reported by Niklas Larsson
Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159
A change to glibc 2.22 changed the order of the sockadddr_storage
members which caused the places where we do an initialization of
ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those
initializers (which we shouldn't have been using anyway) have been
replaced with memsets.
Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4
A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not
loaded and does not have a configuration file. Previously when this
occurred, checks were put in to see if the configuration was loaded
successfully. While this is a good idea - and has been added to the
offending function in res_hep - the reality is res_hep_pjsip and
res_hep_rtcp have no business running if res_hep isn't also running.
As such, this patch also adds a function to res_hep that returns whether
or not it successfully loaded. Oddly enough, ast_module_check returns
"everything is peachy" even if a module declined its load - so it cannot
be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this
function to see if they should continue to load; if it fails, they
decline their load as well.
ASTERISK-26096 #close
Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea
With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", the maximum size of a
single file was shown. Now, the maximum number of possible file descriptors is
shown.
ASTERISK-26097
Change-Id: Icf98d145774b38cac144ca76d19eaef42ce659a3
POSIX defines poll.h, sys/poll.h should not be used at is c-library
internal header which may or may not exist. Notable in musl it
generates warning of being incorrect. And add explict include of
sys/cdefs.h where needed.
Change-Id: I142930df53fe7585a06b854b6faddc5301e024be
When taskprocessors get backed up, there is a good chance that we are
being overloaded and need to defer adding new work to the system.
* Implemented a high/low water alert mechanism for modules to check if the
system is being overloaded and take appropriate action. When a
taskprocessor is created it has default congestion levels set. A
taskprocessor can later have those congestion levels altered for specific
needs if stress testing shows that the taskprocessor is a symptom of
overloading or needs to handle bursty activity without triggering an
overload alert.
* Add CLI "core show taskprocessor" low/high water columns.
* Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was
never a good thing to use when creating a taskprocessor because of the
nature of how its references needed to be cleaned up on a partial
creation.
* Made res_pjsip's distributor check if the taskprocessor overload alert
is active before placing a message representing brand new work onto a
distributor serializer.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I182f1be603529cd665958661c4c05ff9901825fa
We must continue using the serializer that the original INVITE came in on
for the dialog. There may be retransmissions already enqueued in the
original serializer that can result in reentrancy and message sequencing
problems.
Outgoing call legs create the pjsip/outsess/<endpoint> serializers for
their dialogs.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc
* Resolves potential reentrancy problems if system restarted in the middle
of subscription message transactions.
* Fixes memory leak recreating persistent subscriptions when the
subscription resource tree could not be created.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be
We must continue using the serializer that the original SUBSCRIBE came in
on for the dialog. There may be retransmissions already enqueued in the
original serializer that can result in reentrancy and message sequencing
problems. The "sip_transaction Unable to register SUBSCRIBE transaction
(key exists)" message is a notable symptom of this issue.
Outgoing subscriptions still create the pjsip/pubsub/<endpoint>
serializers for their dialogs.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0
Incoming messages that are not part of a dialog or a recognized response
to one of our requests need to be sent to a consistent serializer. Under
load we may be queueing retransmissions before we can process the original
message. We don't need to throw these messages onto random serializers
and cause reentrancy and message sequencing problems.
* Created a pool of pjsip/distributor serializers that get picked by
hashing the call-id and remote tag strings of the received messages.
* Made ast_sip_destroy_distributor() destroy items in the reverse order of
creation.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I2ce769389fc060d9f379977f559026fbcb632407
We should not be processing any incoming messages until we are fully
booted. We may not have dialplan or other needed configuration loaded
yet.
ASTERISK-26089 #close
Reported by: Scott Griepentrog
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264
Testing has shown that our usage of UnixODBC is problematic
due to bugs within UnixODBC itself as well as the heavy weight
cost of connecting and disconnecting database connections, even
when pooling is enabled.
For users of UnixODBC 2.3.1 and earlier crashes would occur due
to insufficient protection of the disconnect operation. This was
fixed in UnixODBC 2.3.2 and above.
For users of UnixODBC 2.3.3 and higher a slow-down would occur
under heavy database use due to repeated connection establishment.
A regression is present where on each connection the database
configuration is cached again, with the cache growing out of
control.
The connection pool implementation present in this change helps
to mitigate these issues by reducing how much we connect and
disconnect database connections. We also solve the issue of
crashes under UnixODBC 2.3.1 by defaulting the maximum number of
connections to 1, returning us to the previous working behavior.
For users who may have a fixed version the maximum concurrent
connection limit can be increased helping with performance.
The connection pool works by keeping a list of active connections.
If the connection limit has not been reached a new connection is
established. If the connection limit has been reached then the
request waits until a connection becomes available before
continuing.
ASTERISK-26074 #close
ASTERISK-26054 #close
Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff
Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore.
Therefore, the symbol RAND_bytes is used instead of crypto_get_random.
ASTERISK-24436 #close
Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96
In several internal library projects, the files are archived with the help of
'ar cr'. Only the projects editline and the Objective Open H.323 stack
implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms
changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier
ignored since `D' is the default (see `U')". For consistency and to avoid this
message all projects use 'ar cr' now.
ASTERISK-26091 #close
Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40
If you create a local channel and don't specify an originator channel
to take capabilities from, we automatically add all audio formats to
the new channel's capabilities. When we try to make the channel
compatible with another, the "best format" functions pick the best
format available, which in this case will be slin192. While this is
great for preserving quality, it's the worst for performance and
overkill for the vast majority of applications.
In the absense of any other information, adding all formats is the
correct thing to do and it's not always possible to supply an
originator so a new parameter 'formats' has been added to the channel
create/originate functions. It's just a comma separated list of formats
to make availalble for the channel. Example: "ulaw,slin,slin16".
'formats' and 'originator' are mutually exclusive.
To facilitate determination of format names, the format name has been
added to "core show codecs".
ASTERISK-26070 #close
Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b
The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting
to use UniqueConstraint and failing. It was not imported and after
importing it also continued to fail.
I've changed the script to use the explicit name of the constraint
instead.
Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9
The stringfields refactor to allow adding stringfields to the end of a
structure (f6f4cf459f) exposed some
incomplete cleanup code by some stringfield users.
The most noticeable leaker is the logging system where there is a leak for
every log message generated.
ASTERISK-26078 #close
Reported by: Etienne Lessard
Patches:
jira_asterisk_26078_v13.patch (license #5621) patch uploaded
by Richard Mudgett
Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782
Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
The pjproject doxygen for rdata->msg_info.info says to call
pjsip_rx_data_get_info() instead of accessing the struct member directly.
You need to call the function mostly because the function will generate
the struct member value if it is not already setup.
Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799
Re-ordered the body items so Message-Account is second.
Messages-Waiting: no
Message-Account: sip:1571@<IP Removed>:5060
Voice-Message: 0/0 (0/0)
ASTERISK-26065 #close
Reported-by: Ross Beer
Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3
Although all the patches we had against 2.4.5 were applied by Teluu,
a new bug was introduced preventing re-use of tcp and tls transports
This patch removes all the previous patches against 2.4.5, updates
the version to 2.5, and adds a new patch to correct the transport
re-use problem.
Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068
Added notes about when you can read or write headers. Specifically
about being able to read on the inbound channel and write on an
outbound channel.
ASTERISK-26063 #close
Reported by: Private Name
Tested by: Rusty Newton
Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5
The pjproject doxygen for rdata->msg_info.info says to call
pjsip_rx_data_get_info() instead of accessing the struct member directly.
You need to call the function mostly because the function will generate
the struct member value if it is not already setup.
Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2
Fixed some bugs:
- create dirpath when save downloading message from IMAP storage.
- create IMAP folder if not exists when saving to IMAP storage
- check if file successfully opened before write to it
- some IMAP checks
- remove non-standard flag 'Unseen'
etc
Change to debug IMAP mm_status log instead of verbose.
Remove unused X-Asterisk-VM-Caller-channel message header
for security reason. The clients should not know name of peer/endpoint.
ASTERISK-26045 #close
Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b
As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.
Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.
ASTERISK-26011
Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
There are a lot of verbose messages about Endpoint and Contact status
changes if there are many dynamic endpoints.
The patch sets verbose level 2 for Endpoint status changes
and verbose level 3 for Contact status changes.
ASTERISK-26055 #close
Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7
worker_start checked for ZOMBIE status without holding a lock. All
other read/write of worker status are performed with a lock, so this
check should do the same.
ASTERISK-25777 #close
Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781
