Richard Mudgett [Thu, 10 Aug 2017 19:18:01 +0000 (14:18 -0500)]
STUN/netsock2: Fix some valgrind uninitialized memory findings.
* netsock2.c: Test the addr->len member first as it may be the only member
initialized in the struct.
* stun.c:ast_stun_handle_packet(): The combinded[] local array could get
used uninitialized by ast_stun_request(). The uninitialized string gets
copied to another location and could overflow the destination memory
buffer.
These valgrind findings were found for ASTERISK_27150 but are not
necessarily a fix for the issue.
Richard Mudgett [Wed, 2 Aug 2017 23:44:12 +0000 (18:44 -0500)]
res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown.
The fix for the issue is broken up into three parts.
This is part three which handles the client side of REGISTER requests.
The registered contact may no longer be valid on the server when the
transport used is reliable and the connection is broken.
* Re-REGISTER our contact if the reliable transport is broken after
registration completes. We attempt to re-REGISTER immediately to minimize
the time we are unreachable. Time may have already passed between the
connection being broken and the loss being detected.
* Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's
are still correct if an allocation failure happens.
Richard Mudgett [Mon, 31 Jul 2017 19:21:06 +0000 (14:21 -0500)]
res_pjsip: Remove ephemeral registered contacts on transport shutdown.
The fix for the issue is broken up into three parts.
This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled. Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.
* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown. If it is shutdown then the contact must be removed because it
is no longer valid. Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there. Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request. The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.
* Prune any rewrite_contact's registered reliable transport contacts on
boot. The reliable transport no longer exists so the contact is invalid.
* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.
* Made the websocket transport set a unique name since that is what we use
as the ao2 container key. Otherwise, we would not know which transport we
find when one of them shuts down. The names are also used for PJPROJECT
debug logging.
* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event. Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.
* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.
Richard Mudgett [Fri, 28 Jul 2017 23:26:17 +0000 (18:26 -0500)]
res_pjsip: PJSIP Transport state monitor refactor.
The fix for the issue is broken up into three parts.
This is part one which refactors the transport state monitor code to allow
more modules to be able to monitor transports.
* Pull the management of PJPROJECT's transport state callback code from
res_pjsip_transport_management.c into res_pjsip. Now other modules can
dynamically add and remove themselves from transport monitoring without
worrying about breaking PJPROJECT's callback chain.
* Add the ability for other modules to get a callback whenever a specific
transport is shutdown.
When handling an incoming SIP MESSAGE, PJSIP
attaches the IP address that the message was
received from to the message in the variable
PJSIP_RECVADDR. When the IP address is IPv6
the :PORT appended results in an unparseable
mess. By using an additional bit flag on the
pj_sockaddr_print call, the conventional use
of brackets around the address is achieved.
George Joseph [Wed, 9 Aug 2017 13:01:33 +0000 (07:01 -0600)]
configure: Add --with-download-cache option
To make building without an internet connection easier, a new
./configure option '--with-download-cache' was added that sets
the cache for externals (like pjproject, the codecs and the DPMA),
AND the sounds files. It can also be specified as an environment
variable named "AST_DOWNLOAD_CACHE". The existing
'--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and
'--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable
remain and if specified, will override '--with-downloads-cache'.
res_rtp_asterisk: enable rtcp & QOS stats on native bridge
Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated. Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.
res_pjsip_session: Release media resources on session end quicker.
A change was made long ago where the session was kept around
until the underlying INVITE session had been destroyed. This
had the side effect of also keeping the underlying media resources
around for this time as well.
This change ensures that when we know the session is ending we
release the media resources immediately.
res_rtp_asterisk: Make P2P bridge Asymmetric codec aware
Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
Corey Farrell [Fri, 4 Aug 2017 02:30:12 +0000 (22:30 -0400)]
Correct some leaks in unit tests.
* chan_sip: channel in test_sip_rtpqos_1.
* test_config: config hook, config info and global config holder.
* test_core_format: format in format_attribute_set_without_interface.
* test_stream: unneeded frame duplication.
* test_taskprocessor: task_data.
res_pjsip_session: Release media resources on session end quicker.
A change was made long ago where the session was kept around
until the underlying INVITE session had been destroyed. This
had the side effect of also keeping the underlying media resources
around for this time as well.
This change ensures that when we know the session is ending we
release the media resources immediately.
Sean Bright [Wed, 26 Jul 2017 13:48:29 +0000 (09:48 -0400)]
res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation
This change fixes PIDF content generation when the underlying device
state is considered in use. Previously it was incorrectly marked
as closed meaning they were offline/unavailable. The code now
correctly marks them as open.
Additionally:
* Generate an XML element for our activity instead of a using a text
node.
* Consider every extension state other than "unavailable" to be 'open'
status.
* Update the XML namespaces and structure to reflect those
documented in RFC 4480
* Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the
"in use" activity. This change results in eyeBeam using the
appropriate icon for the watched user.
This was tested on eyeBeam 1.5.20.2 build 59030 on Windows.
ASTERISK-26659 #close
Reported by: Abraham Liebsch
patches:
ASTERISK-26659.diff submitted by snuffy (license 5024)
res_pjsip: Add support for dnsmgr to external_media_address.
The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
George Joseph [Thu, 27 Jul 2017 11:35:51 +0000 (05:35 -0600)]
bundled_pjproject: Improve SSL/TLS error handling
OpenSSL has 2 levels or error processing. It's possible for the
top layer to return SSL_ERROR_SYSCALL but the lower layer return
no error, in which case processing should continue. Only the top
layer was being examined though so connections were being torn
down when they didn't need to be. This patch adds the examination
of the lower level codes, and if they return no errors, allows
processing to continue.
ASTERISK-27001 Reported-by: Ian Gilmour
patches:
pjproject-2.6.patch submitted by Ian Gilmour (license 6889)
Updated-by: George Joseph and Sauw Ming (Teluu)
Merged to upstream pjproject on 7/27/2017 (commit 5631)
Rusty Newton [Fri, 21 Jul 2017 22:04:31 +0000 (17:04 -0500)]
say.c: Fix file locations for second, seconds, minute, minutes files
The seconds and minutes files have always existed in the base language
directory of the Core package. So say.c has always been calling the wrong
location (under digits/) for those two files and in the case of second and
minute they didn't exist in the Core packages at all.
The 1.6 sounds release moves the second and minute files into Core from
Extra for the languages that already had them. A future release will include
the second and minute files for languages that didn't already have them.
This patch just changes all the target locations for second, seconds,
minute, and minutes that were under the digits subdir to be under the root of
sounds instead. Which is where the sounds will be for some languages after 1.6
sounds and for all languages after a future release.
ASTERISK-25810 #close
Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 Reported-by: Nicolas Riendeau
The tests were noting the start of the vm-incorrect-mailbox prompt and
immediately sending the mailbox for the next login attempt. Since the
invalid message playback had to complete before the digits were
recognized, the test passed for the wrong reason and added approximately
20 seconds to the test times.
* Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox
digits like the initial vm-login prompt so the tests are able to enter the
intended mailbox.
Rusty Newton [Fri, 21 Jul 2017 19:20:10 +0000 (14:20 -0500)]
Sounds: Update Makefile for Extra sounds 1.5.1 release
Incrementing version for the Extra sounds release. 1.5.1 Extra sounds
removes two prompts that were moved into the Core packages in the 1.6 Core
sounds release.
res/res_stasis_snoop: generate silence when audiohook returns null
Currently when rtp is paused, no packets are written to the
recorded audio file, causing the silence to be skipped and recording
not properly time aligned. The read handler as been adapted to
return a silence frame of the correct size.
app_confbridge: Make sure name recordings are always removed from the filesystem
This commit fixes two possible scenarios:
* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.
core: Add PARSE_TIMELEN support to ast_parse_arg and ACO.
This adds support for parsing timelen values from config files. This
includes support for all flags which apply to PARSE_INT32. Support for
this parser is added to ACO via the OPT_TIMELEN_T option type.
Fixes an issue where extra characters provided to ast_app_parse_timelen
were ignored, they now cause an error.
George Joseph [Tue, 11 Jul 2017 12:26:27 +0000 (06:26 -0600)]
res_musiconhold: Add kill_escalation_delay, kill_method to class
By default, when res_musiconhold reloads or unloads, it sends a HUP
signal to custom applications (and all descendants), waits 100ms,
then sends a TERM signal, waits 100ms, then finally sends a KILL
signal. An application which is interacting with an external
device and/or spawns children of its own may not be able to exit
cleanly in the default times, expecially if sent a KILL signal, or
if it's children are getting signals directly from
res_musiconhoild.
* To allow extra time, the 'kill_escalation_delay'
class option can be used to set the number of milliseconds
res_musiconhold waits before escalating kill signals, with the
default being the current 100ms.
* To control to whom the signals are sent, the "kill_method" class
option can be set to "process_group" (the default, existing
behavior), which sends signals to the application and its
descendants directly, or "process" which sends signals only to the
application itself.
George Joseph [Wed, 5 Jul 2017 20:31:43 +0000 (14:31 -0600)]
http.c: Reduce log spam
Messages like "fwrite() failed: Connection reset by peer" are no
help whatsoever, especially since they can be caused simply by a
client disconnecting.
* Make those WARNINGs DEBUGs.
* Check the return of the headers fprintf.
res_pjsip: Fix crash with from_user containing invalid characters.
If the from_user field contains certain characters (like @, {, ^, etc.),
PJSIP will return a null value for the URI when attempting to parse it.
This causes a crash when trying to dial out through a trunk that contains
these invalid characters in its from_user field.
This change checks the configuration and ensures that an endpoint will
not be created if the from_user contains an invalid character. It also
adds a null check to the PJSIP URI parsing as a backup.
Richard Mudgett [Wed, 5 Jul 2017 18:39:45 +0000 (13:39 -0500)]
res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock.
When a message is received on the TURN socket, the code processing the
message needs to call into the ICE/STUN session for further processing.
This code path locks the TURN group lock then the ICE/STUN group lock. In
another thread an ICE/STUN timer can fire off to send a keep alive message
over the TURN socket. In this code path, the ICE/STUN group lock is
obtained then the TURN group lock is obtained to send the packet. A
classic deadlock case if the group locks are not the same.
* Made TURN get created using the ICE/STUN session's group lock.
NOTE: I was originally concerned that the ICE/STUN session can get
recreated by ice_reset_session() for an event like RTCP multiplexing
causing a change during SDP negotiation. In this case the TURN group lock
would become different. However, TURN is also recreated as part of the
ICE/STUN recreation in ice_create() when all known ICE candidates are
added to the new ICE session. While the ICE/STUN and TURN sessions are
being recreated there is a period where the group locks could be
different.
ASTERISK-27023 #close
Patches:
res_rtp_asterisk-turn-deadlock-fix.patch (license #6502)
patch uploaded by Michael Walton (modified)
Richard Mudgett [Fri, 23 Jun 2017 16:17:51 +0000 (11:17 -0500)]
bridge_native_rtp.c: Fix direct media video RTP instance ACL check.
The video stream was using the audio stream RTP instance addresses to
check if the video RTP gets directed to an allowed direct media Access
Control List (ACL) address. There is no guarantee that the video RTP
instance uses the same addresses as the audio RTP instance.
This looks like it has been a bug since v11 when direct media ACL was
first added to chan_sip and then faithfully reproduced through a couple
code refactorings into the new bridging architecture.
Sean Bright [Wed, 5 Jul 2017 12:42:07 +0000 (08:42 -0400)]
core: Fix segfault when invoking 'data get' CLI command
Invoking 'data get /asterisk/core/channeltypes' caused a crash because
of an assumption of a tech's capabilities to be non-NULL. The
'Surrogate' tech, however, does have a NULL capabilities member,
resulting in a crash.
Alexander Traud [Mon, 3 Jul 2017 15:59:43 +0000 (17:59 +0200)]
chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).
When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was
added in any case, because of a local Boolean-negation error of the return value
of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was
still always added with tlsenable=yes, because the domains were not compared
just on the address but also on the port – and TLS is always on a different port
than UDP/TCP.
Alexander Traud [Mon, 3 Jul 2017 15:38:32 +0000 (17:38 +0200)]
chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).
Because of a copy-and-paste error when the struct ast_sockaddr changed,
tlsbindaddr was not added, when sip.conf contained autodomain=yes; see
"show sip domains" on the command-line interface (CLI) of Asterisk.
projects / thirdparty / asterisk.git / log
