Ben Ford [Wed, 28 Aug 2019 19:25:57 +0000 (14:25 -0500)]
res_rtp: Add unit tests for RTCP stats.
Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.
While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.
Also made a minor fix to the data buffer unit test.
George Joseph [Mon, 5 Aug 2019 11:59:59 +0000 (05:59 -0600)]
ARI: External Media
The Channel resource has a new sub-resource "externalMedia".
This allows an application to create a channel for the sole purpose
of exchanging media with an external server. Once created, this
channel could be placed into a bridge with existing channels to
allow the external server to inject audio into the bridge or
receive audio from the bridge.
See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
for more information.
Joshua Colp [Mon, 26 Aug 2019 12:53:27 +0000 (09:53 -0300)]
AST-2019-005 - translate: Don't assume all frames will have a src.
This change removes the assumption that a frame will always have
a src set on it. This assumption is incorrect.
Given a scenario where an RTP packet is received with no payload
the resulting audio frame will have no samples. If this frame goes
through a signed linear translation path an interpolated frame can
be created (if generic packet loss concealment is enabled) that has
minimal data on it, including no src. If this frame is given to a
translation path a crash will occur due to the lack of src.
Kevin Harwell [Tue, 20 Aug 2019 20:05:45 +0000 (15:05 -0500)]
AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media
After receiving a 200 OK with a declined stream in response to a T.38
initiated re-invite Asterisk would crash when attempting to dereference
a NULL session media object.
This patch checks to make sure the session media object is not NULL before
attempting to use it.
ASTERISK-28495
patches:
ast-2019-004.patch submitted by Alexei Gradinari (license 5691)
test_utils.c: Skip test adsi_loaded_test if module not loaded.
Module res_adsi.so is deprecated, therefore it does not load by default.
Module not loaded causes it to yield a FAIL when tested by tests/test_utils.c.
This fix checks if the corresponding module is loaded at the start of the test,
and if not, it passes the test and exits with a message.
This fix is applied to all versions where the module is marked deprecated.
chan_unistim: Fix clang warning: variable sized type not at end of a struct
On reading information about initial client packet unistim use dirty
implementation of destination ip address retrieval. This fix uses
CMSG_*(..) to get ip address and make clang compile without warning.
ASTERISK-25592 #close Reported-by: Alexander Traud
Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1
Kevin Harwell [Fri, 23 Aug 2019 22:03:07 +0000 (17:03 -0500)]
res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions
res_pjsip_mwi allows both solicited and unsolicited MWI subscription types.
While both can be set in the configuration for a given endpoint/aor, only
one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor
is configured to allow both types then the solicited subscription is rejected
when it comes in. However, there is a configuration option to override that
behavior:
mwi_subscribe_replaces_unsolicited
When set to "yes" then when a solicited subscription comes in instead of
rejecting it Asterisk is suppose to replace the unsolicited one if it exists.
Prior to this patch there was a bug in Asterisk that allowed the solicted one
to be added, but did not remove the unsolicited. As a matter of fact a new
unsolicited subscription got added everytime a SIP register was received.
Over time this eventually could "flood" a phone with SIP notifies.
This patch fixes that behavior to now make it work as expected. If configured
to do so a solicited subscription now properly replaces the unsolicited one.
As well when an unsubscribe is received the unsolicited subscription is
restored. Logic was also put in to handle reloads, and any configuration changes
that might result from that. For instance, if a solicited subscription had
previously replaced an unsolicited one, but after reload it was configured to
not allow that then the solicited one needs to be shutdown, and the unsolicited
one added.
chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk
Current implementation of ast_channel_tech send_digit_begin hook uses
same function for tone playback as key press handler. This cause every
incoming dtmf send back to asterisk. In case of two unistim phones
connected to each other, it'll cause indefinite DTMF loop. Fix add
separate function for dtmf tone phone play.
chan_unistim: Fix RTP port byte order for big-endian arch
This patch fixes one-way oudio that users expirienced on
big-endian architechtires. RTP port number bytes was stored
in improper order and phone sent RTP to wrong RTP port.
George Joseph [Wed, 21 Aug 2019 18:29:57 +0000 (12:29 -0600)]
chan_rtp: Accept hostname as well as ip address as destination
The UnicastRTP channel driver provided by chan_rtp now accepts
"<hostname>:<port>" as an alternative to "<ip_address>:<port>"
in the destination. The first AAAA (preferred) or A record resolved
will be used as the destination. The lookup is synchronous so beware
of possible dialplan delays if you specify a hostname.
George Joseph [Wed, 21 Aug 2019 17:03:26 +0000 (11:03 -0600)]
dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4
The new function takes in a pointer to an ast_sockaddr structure,
a hostname and an optional port and then dispatches parallel
"AAAA" and "A" record queries. If an "AAAA" record is returned,
it's parsed into the ast_sockaddr structure along with the port
if it was supplied. If no "AAAA" record was returned, the
first "A" record returned (if any) is parsed instead.
This is a synchronous call. If you need asynchronous lookups,
use ast_dns_query_set_resolve_async and roll your own.
Dan Cropp [Wed, 21 Aug 2019 15:58:00 +0000 (10:58 -0500)]
pjproject: Configurable setting for cnonce to include hyphens or not
NEC SIP Station interface with authenticated registration only supports cnonce
up to 32 characters. In Linux, PJSIP would generate 36 character cnonce
which included hyphens. Teluu developed this patch adding a compile time
setting to default to not include the hyphens. They felt it best to still
generate the UUID and strip the hyphens.
They have indicated it will be part of PJSIP 2.10.
ASTERISK-28509 Reported-by: Dan Cropp
Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470
In chan_sip, there was variable SIPFROMDOMAIN that allows to set
From header URI domain per channel. This patch introduces res_pjsip
variable SIPFROMDOMAIN for backward compatibility with chan_sip.
George Joseph [Tue, 20 Aug 2019 18:04:56 +0000 (12:04 -0600)]
res_ari.c: Prefer exact handler match over wildcard
Given the following request path and 2 handler paths...
Request: /channels/externalMedia
Handler: /channels/{channelId} "wildcard"
Handler: /channels/externalmedia "non-wildcard"
...if /channels/externalMedia was registered as a handler after
/channels/{channelId} as shown above, the request would automatically
match the wildcard handler and attempt to parse "externalMedia" into
the channelId variable which isn't what was intended. It'd work
if the non-wildard entry was defined in rest-api/api-docs/channels.json
before the wildcard entry but that makes the json files
order-dependent which isn't a good thing.
To combat this issue, the search loop saves any wildcard match but
continues looking for exact matches at the same level. If it finds
one, it's used. If it hasn't found an exact match at the end of
the current level, the wildcard is used. Regardless, after
searching the current level, the wildcard is cleared so it won't
accidentally match for a different object or a higher level.
BTW, it's currently not possible for more than 1 wildcard entry
to be defined for a level. For instance, there couldn't be:
Handler: /channels/{channelId}
Handler: /channels/{channelName}
We wouldn't know which one to match.
Sean Bright [Fri, 9 Aug 2019 20:53:03 +0000 (16:53 -0400)]
audiohook.c: Substitute silence for unavailable audio frames
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Alexei Gradinari [Wed, 14 Aug 2019 19:52:01 +0000 (15:52 -0400)]
app_voicemail/IMAP: check mailstream not NULL in leave_voicemail
The function leave_voicemail checks if expungeonhangup is set,
but does not check if IMAP stream is closed,
so it could call imap function with NULL stream.
This leads to segfault.
Kevin Harwell [Wed, 7 Aug 2019 22:54:34 +0000 (17:54 -0500)]
srtp: Fix possible race condition, and add NULL checks
Somehow it's possible for the srtp session object to be NULL even though the
Asterisk srtp object itself is valid. When this happened it would cause a
crash down in the srtp code when attempting to protect or unprotect data.
After looking at the code there is at least one spot that makes this situation
possible. If Asterisk fails to unprotect the data, and after several retries
it still can't then the srtp->session gets freed, and set to NULL while still
leaving the Asterisk srtp object around. However, according to the original
issue reporter this does not appear to be their situation since they found
no errors logged stating the above happened (which Asterisk does for that
situation).
An issue was found however, where a possible race condition could occur between
the pjsip incoming negotiation, and the receiving of RTP packets. Both places
could attempt to create/setup srtp for the same rtp instance at the same time.
This potentially could be the cause of the problem as well.
Given the above this patch adds locking around srtp setup for a given rtp, or
rtcp instance. NULL checks for the session have also been added within the
protect and unprotect functions as a precaution. These checks should at least
stop Asterisk from crashing if it gets in this situation again.
This patch also fixes one other issue noticed during investigation. When doing
a replace the old object was freed before creating the replacement. If the new
replacement object failed to create then the rtp/rtcp instance would now point
to freed srtp data which could potentially cause a crash as well when the next
attempt to reference it was made. This is now fixed so the old srtp object is
kept upon replacement failure.
Lastly, more logging has been added to help diagnose future issues.
George Joseph [Thu, 8 Aug 2019 12:12:18 +0000 (06:12 -0600)]
CI: Add "throttle" label and "skip_gate" capability
To make throttling by label fully active, the "throttle" option
has to be specified with a specific label.
You can now specify "skip_gate" in the Gerrit comments when you
do a +2 code review to tell Jenkins not to actually run the
gate. You'd do this if you plan to manually merge the change.
Also updated the "printenv" debug output to better sort multi-line
comments.
Joshua Colp [Mon, 5 Aug 2019 12:23:53 +0000 (09:23 -0300)]
cdr / cel: Use event time at event creation instead of processing.
When updating times on CDR or CEL records using the time at which
it is done can result in times being incorrect if the system is
heavily loaded and stasis message processing is delayed.
This change instead makes it so CDR and CEL use the time at which
the stasis messages that drive the systems are created. This allows
them to be backed up while still producing correct records.
George Joseph [Tue, 6 Aug 2019 15:40:54 +0000 (09:40 -0600)]
CI: Make node labels job-specific
Originally, the eligible nodes for a job were labelled only by
"swdev-docker". So basically any node could run any job. We had
found that allowing a node to run more than 1 gate at a time was
problematic so we limited the nodes to processing 1 job at a time.
With the creation of the Asterisk 17 branches however, we now have
so many active branches that getting checks and gates through in
a timely manner is problematic when a node can run only 1 job
at a time.
Now the nodes are also labelled by the job type they can run.
For instance: "asterisk-check", "asterisk-gate", etc. With the
"Throttle Concurrent Builds" plugin, we can now allow a node to
run more than 1 job BUT throttle by job type. For instance:
Allow 2 jobs but only 1 asterisk-gate at a time.
Now a node can run 2 checks or 1 check and 1 gate or 1 gate but
not 2 gates at a time.
Kevin Harwell [Thu, 1 Aug 2019 21:22:01 +0000 (16:22 -0500)]
various modules: json integer overflow
There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:
unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);
would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.
Joshua Colp [Thu, 1 Aug 2019 10:07:45 +0000 (10:07 +0000)]
res_pjsip: Fix multiple of the same contact in "pjsip show contacts".
The code for gathering contacts could result in the same contact
being retrieved and added to the list multiple times. The container
which stores the contacts to display will now only allow a contact
to be added to it once instead of multiple times.
Sean Bright [Wed, 24 Jul 2019 20:12:49 +0000 (16:12 -0400)]
manager: Send fewer packets
The functions that build manager message headers do so in a way that
results in a single messages being split across multiple packets. While
this doesn't matter to the remote end, it makes network captures noisier
and harder to follow, and also means additional system calls.
With this patch, we build up more of the message content into the TLS
buffer before flushing to the network. This change is completely
internal to the manager code and does not affect any of the existing
API's consumers.
George Joseph [Mon, 29 Jul 2019 13:31:56 +0000 (07:31 -0600)]
loader.c: Fix possible SEGV when a module fails to register
When a module fails to register itself (usually a coding error
in the module), dlerror() can return NULL. We weren't checking
for that in load_dlopen() before trying to strdup the error message
so a SEGV was thrown. dlerror() is now surrounded with an S_OR
so we don't SEGV.
George Joseph [Wed, 24 Jul 2019 20:15:27 +0000 (14:15 -0600)]
CI: Don't enable non-core modules in Certified branches
We don't support non-core modules for Certified releases but we
were enabling them for CI builds which was causing lots of test
failures. Now we don't.
Details:
- The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2"
- As a result, the memcmp() in dahdi_r2_get_link() always fails
- This cause dahdi_r2_get_link() to create new link for every channel
(instead of a new link for every ~30 channels)
- With the fix, far less links are generated -- so we use far less threads
George Joseph [Fri, 19 Jul 2019 16:20:38 +0000 (10:20 -0600)]
CI: Add cleanWs to cleanup steps in jenkinsfiles
We're at the point where there are enough Jenkins jobs for
Asterisk branches than even cleaned checkouts of Asterisk
will add up to more disk space than is available on the
in-memory workspace mount. Since we archive all relevent
artifacts anyway, there's no need to keep the workspace
around after the job finishes, whether it succeeds or fails.
Walter Doekes [Wed, 17 Jul 2019 13:06:12 +0000 (15:06 +0200)]
sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread
When fixing ASTERISK~24212, a change was done so a scheduled callback could not
be removed while it was running. The caller of ast_sched_del would have to wait.
However, when the caller of ast_sched_del is the callback itself (however wrong
this might be), this new check would cause a deadlock: it would wait forever
for itself.
This changeset introduces an additional check: if ast_sched_del is called
by the callback itself, it is immediately rejected (along with an ERROR log and
a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the
after-ast_sched_del-refcall function is only run if ast_sched_del returned
success.
This should fix the following spurious race condition found in chan_sip:
- thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE)
- thread 2: run sip_poke_peer_now
- thread 2: blank out sched-ID (too soon!)
- thread 1: set sched-ID (too late!)
- thread 2: try to delete the currently running sched-ID
After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor
excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and
other madness) should occur.
(Thanks Richard Mudgett for reviewing/improving this "scary" change.)
Note that this change does not fix the observed race condition: unlocked
access to peer->pokeexpire (and potentially other scheduled items in chan_sip),
causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the
deadlock go away. And in the observed case, it will not have adverse affects
(like memory leaks) because the scheduled item is removed through a different
path.
George Joseph [Tue, 16 Jul 2019 12:55:49 +0000 (06:55 -0600)]
Build: Separate header install/uninstall
Asterisk headers are no longer installed and uninstalled
automatically when performing a "make install" or a
"make uninstall". To install/uninstall the headers, use
"make install-headers" and "make uninstall-headers".
The headers also continue to be uninstalled when performing a
"make uninstall-all".
Also corrects an issue where /usr/include/asterisk.h was never
being removed at all.
chan_sip: Handle invalid SDP answer to T.38 re-invite
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.
If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.
Kevin Harwell [Wed, 12 Jun 2019 18:49:30 +0000 (13:49 -0500)]
app_voicemail: Remove dependency on the stasis cache
app_voicemail utilized the stasis cache when polling mailboxes for MWI. This
caused a memory leak (items were not being appropriately removed from the
cache), and subsequent slowdown in system processing. This patch removes the
stasis cache dependency, thus alleviating the memory leak. It does this by
utilizing the new MWI API that better manages state lifetime.
Kevin Harwell [Wed, 12 Jun 2019 18:11:42 +0000 (13:11 -0500)]
MWI: Update modules that subscribe to MWI to use new API calls
The MWI core recently got some new API calls that make tracking MWI state
lifetime more reliable. This patch updates those modules that subscribe to
specific MWI topics to use the new API. Specifically, these modules now
subscribe to both MWI topics and MWI state.
Kevin Harwell [Tue, 11 Jun 2019 19:12:12 +0000 (14:12 -0500)]
mwi: Update the MWI core to use stasis_state API
** Note **
This patch is meant to be the minimum needed in order for the MWI core to use
the now underlying stasis_state module. As such it does not completely remove
its reliance on the stasis_cache. Doing so has allowed current consumers to
not have to change, and update those code paths for this patch. When time
allows, subsequent patches can/will be made to those consumers to take advantage
of some of the new MWI API included here. Thus, eventually and ultimately
removing MWI dependency on the stasis_cache.
** End Note **
This patch makes it so the MWI core now takes advantage of the new stasis_state
API. Consumers of MWI should no longer need to depend upon stasis topic pooling,
and the stasis cache directly. Similar functionality and implementation details
have now been pushed into the stasis_state module. However, all MWI state should
be accessed via the MWI API itself.
As such a few new methods, and constructs have been added to the MWI core that
facilitate consumer publishing, subscribing, and iterating over MWI state data.
* ast_mwi_subscriber *
Created via ast_mwi_add_subscriber, a subscriber subscribes to a given mailbox
in order to receive updates about the given mailbox. Adding a subscriber will
create the underlying topic, and associated state data if those do not already
exist for it. The topic, and last known state data is guaranteed to exist for
the lifetime of the subscriber.
* ast_mwi_publisher *
Before publishing to a particular topic a publisher should be created. This can
be achieved by using ast_mwi_add_publisher. Publishing to a mailbox should then
be done using one of the MWI publish functions. This ensures the message is
published to the appropriate topic, and the last known state is maintained.
* ast_mwi_observer *
Add an observer in order to watch for particular MWI module related events. For
instance if a submodule needs to know when a subscription is added to any
mailbox an observer can be added to watch for that.
* other *
Urgent message count is now part of the published MWI state object. Also state
can be iterated over using defined callbacks.
Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous
patch introduced a variable of type unassigned long long which is 64-bits.
Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work
with 32-bit systems.
projects / thirdparty / asterisk.git / log
