res_pjsip: Change log message from error to warning for valid use cases
If a SIP MESSAGE is triggered for an endpoint that is currently not registered
- and therefore has no valid contact associated - an error message was logged.
Since this is a valid request in a valid use cases this is now changed to a
warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list.
George Joseph [Fri, 20 Jul 2018 11:20:25 +0000 (05:20 -0600)]
xmldoc.c: Fix dump of xml document
The "xmldoc dump" cli command was simply concatenating xml documents
into the output file. The resulting file had multiple "xml"
processing instructions and multiple root elements which is illegal.
Normally this isn't an issue because Asterisk has only 1 main xml
documentation file but codec_opus has its own file so if it's
downloaded and you do "xmldoc dump", the result is invalid.
* Added 2 new functions to xml.c:
ast_xml_copy_node_list creates a copy of a list of children.
ast_xml_add_child_list adds a list to an existing list.
* Modified handle_dump_docs to create a new output document and
add to it the children from each input file. It then dumps the
new document to the output file.
build_tools/make_version: Get MAINLINE_BRANCH from .gitreview.
Use .gitreview defaultbranch setting to determine the mainline branch.
This allows the script to be used against other directories which might
not be on the same defaultbranch. This can be used by CI scripts to
report the testsuite version being used:
./build_tools/make_version ${TESTSUITE_DIR}
sched: Make ABI compatible between dev mode and non-dev mode.
In the past there was an assertion in the ast_sched_del function
and in order to ensure it was useful the calling function name,
line number, and filename had to be passed in. This cause the ABI
to be different between dev mode and non-dev mode.
This assertion is no longer present so the special logic can be
removed to make it the same between them both.
res_pjsip: Update default keepalive interval to 90 seconds.
A change recently went in which disabled the built-in PJSIP
keepalive. This defaulted to 90 seconds and kept TCP/TLS
connections alive. Disabling this functionality has resulted
in a behavior change of not doing keepalives by default resulting
in TCP/TLS connections dropping for some people.
This change makes our default keepalive interval 90 seconds
to match the previous behavior and preserve it.
Previously, Asterisk did not tell its bundled PJProject about this configure
parameter. Therefore, PJProject used the platform provided OpenSSL always.
Ben Ford [Thu, 10 May 2018 18:11:06 +0000 (13:11 -0500)]
res_rtp_asterisk: Add support for sending NACK requests.
Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.
If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.
According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.
Also added additional functionality to ast_data_buffer, along with some
testing.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
res_sorcery_config: Allow configuration section to be used based on name.
A problem I've seen countless times is a global or system section
for PJSIP not getting applied. This is inevitably the result of
the "type=" line missing. This change alleviates that problem.
The ability to specify an explicit section name has been
added to res_sorcery_config. If the configured section
name matches this and there are no unknown things configured
the section is taken as being for the given type.
Both the PJSIP "global" and "system" types now support this
so you can just name your section "global" or "system" and it
will be matched and used, even without a "type=" line.
stasis: Improve message type "Use of before/init after destruction"
Fixes issue where error msg
"Use of before/init after destruction"
was being printed on disabled messages
in dev mode. With this
fix if message is disabled
a warning will print.
Nick French [Tue, 17 Jul 2018 14:09:04 +0000 (09:09 -0500)]
res_pjsip: Remove spurious error logging when printing silent headers
Asterisk patched the pjproject source to avoid crashing when pjproject
sip_msg headers are encountered with NULL vptr's, but the patch also
output error messages for some valid headers which simply did not need
to be added to the message itself, such as hidden route headers.
pjproject has since applied a similar patch to their baseline to avoid
crashes, but their version also avoids the spurious error logging.
* Merge the preload and load stages, use load ordering to try preload's
first. This fixes an issue where `preload=res_config_curl` would fail
unless res_curl and func_curl were also preloaded. Now it is only
required that those modules be loaded during startup: autoload or
regular load is good enough.
* The configuration option `require` and `preload-require` were only
effective if the modules failed to load. These options will now abort
Asterisk startup if required modules fail to reach the 'Running'
state.
* Missing or invalid 'module.conf' did not prevent startup. Asterisk
doesn't do anything without modules so this a fatal error.
Richard Mudgett [Fri, 13 Jul 2018 23:26:46 +0000 (18:26 -0500)]
Build: Fix modules getting their optimization setting overridden.
Asterisk modules that use PJPROJECT services have their compiler
optimization and possibly their symbolic debug options overridden by the
PJPROJECT configure script selected settings.
* We need to filter-out any -O and -g options in PJ_CFLAGS before echoing
out the result so the PJPROJECT_INCLUDE variable does not override the
Asterisk module settings when using bundled PJPROJECT.
NOTE: This patch only has an effect when using bundled PJPROJECT.
George Joseph [Tue, 10 Jul 2018 18:28:09 +0000 (12:28 -0600)]
app_confbridge: Use the SDP 'label' attribute to correlate users
Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.
Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label"
attribute to the current sdp.
Turn off the periodic sending of CRLNCRLN. Default is on (90 seconds),
which conflicts with the global section's keep_alive_interval option in
pjsip.conf.
patches:
pjsip_keep_not_alive.patch submitted by Alexander Traud (License 6520)
George Joseph [Wed, 11 Jul 2018 11:14:49 +0000 (05:14 -0600)]
CI: Initial commit for moving CI into source repo
Create tests/CI directory and add files used by Jenkins to
build and test Asterisk.
With this commit, Jenkins will run the Asterisk Unit Tests using
the Jenkinsfile at tests/CI/unittests.jenkinsfile. Bash scripts
to do the actual building and testing are also in the same directory.
Output is placed in tests/CI/output so that directory has been
added to .gitignore.
projects / thirdparty / asterisk.git / log
