Prometheus is the defacto monitoring tool for containerized applications.
This patch adds native support to Asterisk for serving up Prometheus
compatible metrics, such that a Prometheus server can scrape an Asterisk
instance in the same fashion as it does other HTTP services.
The core module in this patch provides an API that future work can build
on top of. The API manages metrics in one of two ways:
(1) Registered metrics. In this particular case, the API assumes that
the metric (either allocated on the stack or on the heap) will have
its value updated by the module registering it at will, and not
just when Prometheus scrapes Asterisk. When a scrape does occur,
the metrics are locked so that the current value can be retrieved.
(2) Scrape callbacks. In this case, the API allows consumers to be
called via a callback function when a Prometheus initiated scrape
occurs. The consumers of the API are responsible for populating
the response to Prometheus themselves, typically using stack
allocated metrics that are then formatted properly into strings
via this module's convenience functions.
These two mechanisms balance the different ways in which information is
generated within Asterisk: some information is generated in a fashion
that makes it appropriate to update the relevant metrics immediately;
some information is better to defer until a Prometheus server asks for
it.
Note that some care has been taken in how metrics are defined to
minimize the impact on performance. Prometheus's metric definition
and its support for nesting metrics based on labels - which are
effectively key/value pairs - can make storage and managing of metrics
somewhat tricky. While a naive approach, where we allow for any number
of labels and perform a lot of heap allocations to manage the information,
would absolutely have worked, this patch instead opts to try to place
as much information in length limited arrays, stack allocations, and
vectors to minimize the performance impacts of scrapes. The author of
this patch has worked on enough systems that were driven to their knees
by poor monitoring implementations to be a bit cautious.
Additionally, this patch only adds support for gauges and counters.
Additional work to add summaries, histograms, and other Prometheus
metric types may add value in the future. This would be of particular
interest if someone wanted to track SIP response types.
Finally, this patch includes unit tests for the core APIs.
ASTERISK-28403
Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address
This causes both 192.168.1.1 and 1.2.3.4 to be advertized.
Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db
The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.
This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".
In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.
ASTERISK-27994 #close
Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73
Various fixes for issues caught by gcc 9. Mostly snprintf
trying to copy to a buffer potentially too small.
ASTERISK-28412
Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e
This change fixes two bugs which both resulted in the packet loss
count exceeding 65,000.
The first issue is that the sequence number check to determine if
cycling had occurred was using the wrong variable resulting in the
check never seeing that cycling has occurred, throwing off the
packet loss calculation. It now uses the correct variable.
The second issue is that the packet loss calculation assumed that
the received number of packets in an interval could never exceed
the expected number. In practice this isn't true due to delayed
or retransmitted packets. The expected will now be updated to
the received number if the received exceeds it.
ASTERISK-28379
Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6
When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.
Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415
Added a conversion for umax (largest maximum sized integer allowed). Adjusted
the other current conversion functions (uint and ulong) to be derivatives of
the umax conversion since they are simply subsets of umax.
Also made the negative check move the pointer on spaces since strtoumax does it
anyways.
Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08
When we use early bridge with create and dial from stasis using Local channel
and the dialplan does not any entry the it is returned from core_local.c with
No such extension .
In such case asterisk locks up till the channel is not hangup with the error
Exceptionally long voice queue length
* Found that in such case app_control_dial fails on ast_call method and
return -1
* Since it is called from stasis_app_send_command_async and return -1 does
not cause resources to be freed and since no PBX exist it is not able to
read from channel causing exceptionally long queue
* After putting this code found that the channel was releasing immediately
and resources were freed.
ASTERISK-28399
Reported by: Abhay Gupta
Tested by: Abhay Gupta
Change-Id: I0a55c923fc6995559f808d63b9488762b4489318
When the gcc version is >= 8.2.1, we were already setting the
--fno-partial-inlining flag for Asterisk source files to get around
a gcc bug but we weren't passing the flag down to the bundled
builds of pjproject and jansson.
ASTERISK-28392
Change-Id: I99ede9bc35408ecd096f7d5369e8192d3dc75704
Updated ast_sip_create_rdata_with_contact and registrar_find_contact
to check the return from pjsip_parse_uri before attempting to
use the uri returned.
ASTERISK-28402
Reported-by: Ross Beer
Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7
The dial bridge is meant to hold channels which have been created
and dialed in stasis. It handles the frames coming from them and raises
the appropriate events.
It was possible for the code to mistakenly place calls which came
from the dialplan into the dial bridge if they were not in an
answered state. These channels are not outgoing channels and
should not be placed into the dial bridge.
The code now checks to ensure that only stasis created channels are
placed into the dial bridge by checking that a PBX does not exist
on the channel.
ASTERISK-27756
Change-Id: Ideee69ff06c9a0b31f7ed61165f5c055f51d21b6
After a bridge has been deleted the stasis control will depart
the channel and might attempt to re-add it to the dial bridge.
The later can fail and this can lead to a situation that the stasis
control is unlinked but the after_bridge_cb_failed cb is executed trying
to access a dangling control object.
Fix it by calling the after_cb's before bridge_channel_impart_signal.
ASTERISK-26718
Change-Id: Ib4e8f70d7a21bd54afe3cb51cc6717ef7c355496
When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.
This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.
ASTERISK-28401
Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c
The transport-cc draft is a mechanism by which additional information
about packet reception can be provided to the sender of packets so
they can do sender side bandwidth estimation. This is accomplished
by having a transport specific sequence number and an RTCP feedback
message. This change implements this in the receiver direction.
For each received RTP packet where transport-cc is negotiated we store
the time at which the RTP packet was received and its sequence number.
At a 1 second interval we go through all packets in that period of time
and use the stored time of each in comparison to its preceding packet to
calculate its delta. This delta information is placed in the RTCP
feedback message, along with indicators for any packets which were not
received.
The browser then uses this information to better estimate available
bandwidth and adjust accordingly. This may result in it lowering the
available send bandwidth or adjusting how "bursty" it can be.
ASTERISK-28400
Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc
There a long history here:
In commit dd1e62c095 has introduce by default shared_lastcall = true by
default but this now only happen is there not [general] directive in
queues.conf
After that, the commit 4b50e3f1ee fix the
sample file.
We'll need to keep the same setting if there a general or not section in
configuration file since the shared_lastcall is by a long time in
sample files as default value to 'no'.
Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c
When compiling in dev mode, stasis statistics are enabled and can cause
a crash at shutdown due to the following:
- Containers are freed
- Topics and subscriptions remain
- When those topics and subscriptions are deallocated, they go to do
things with the container
This changes the containers to global ao2 objects, and whenever needed
in the code, a reference must be obtained and checked before any
operations can be done.
ASTERISK-28353 #close
Change-Id: Ie7d5e907fcfcb4d65bd36d5e4eb923126fde8d33
Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
ASTERISK-28363
Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0
Make compiler check use the output of the actual compiler being
used as reported by the CC variable, instead of unconditionally
running the "gcc" binary. Also only run the check if the compiler
is gcc or a cross-compile gcc.
ASTERISK-28374
Change-Id: Icaacf6d93686ad21076878aa1504a23b4fc9d0f4
We changed the validation of autocomplete parameter in the "indications
remove" command to avoid continue the execution of the command after
asking for autocomplete out of range parameters.
ASTERISK-28391
Reported by: lmendes86
Change-Id: I92b24131fd02f2e3c7fec966eea6f7a663310d40
Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.
res_pjsip_refer: Configures PJSIP norefersub capability accordingly.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close
Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
Currently, the "stasis show app" cli doesn't give detail
of subscription/subscriber information.
Added more printings to show details.
ASTERISK-28378
Change-Id: If25a6f14fe4f622bfb37462e891333da1fdf875f
