Prior to this patch, local candidate lists including SRFLX would fail to start
properly when building ICE candidate check lists. This patch fixes that problem
by making sure that each SRFLX candidate is associated with the proper
base address so that the check list can create matches properly.
This patch was written by jcolp. The issue will be left open to await testing
by the issue participants.
(issue ASTERISK-23213)
Reported by: Andrea Suisani
Review: https://reviewboard.asterisk.org/r/3256/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change minrate from 2400 to 4800 on config reload in response to changes from
ASTERISK-22790 only. Any config with minrate of 2400 that would fail before
r405693 will still fail.
Comment out many settings in res_fax.conf.sample. The defaults are set in
res_fax.c, so setting the same value in sample config does nothing but make
the sample config more fragile.
(closes issue ASTERISK-23231)
Reported by: David Brillert
Review: https://reviewboard.asterisk.org/r/3261/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.
The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.
(closes issue ASTERISK-19773)
Reported by: Joel Vandal
(closes issue ASTERISK-22757)
Reported by: Gareth Blades
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_bind to a port reserved for another program by SELinux causes
errno == EACCES. This caused random failures when binding rtp or
udptl sockets. Treat EACCES as a non-fatal error, try next port.
(closes issue ASTERISK-23134)
Reported by: Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This restricts direct usage of global oseq so that all accesses are
locked and threads are not racing to get oseq values that they did not
claim.
This also fixes a build error in res_pktccops under dev mode.
(closes issue ASTERISK-23100)
Reported by: adomjan
Patch by: adomjan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
According to the new standard for V.27 and V.32 they are able to transmit
at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be
updated to reflect this. Also, because of this change the default 'minrate'
value was updated to be 4800.
(closes issue ASTERISK-22790)
Reported by: Paolo Compagnini
Patches:
res_fax.txt uploaded by looserouting (license 6548)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A 'make distclean' is equivalent to 'make dist-clean' in the top most Makefile.
This patch updates the res/Makefile to recognize both distclean and dist-clean.
Note that this is needed for removing build.mak, which can run into problems
if the source file of Asterisk or its path is changed after build.mak is
generated.
(issue ASTERISK-22480)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_rtp_ice_start if the ice session create check list failed, start check
was never initiated and ice_started was never set to true. Upon re-entering
the function (for instance, [un]hold) it would try to create the check list
again with duplicate remote candidates.
Fixed so that if the create check list fails the necessary data structures
are properly re-initialized for any subsequent retries.
Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a
check list failure because it possible things might still work. However, a
debug message was added to help with any future troubleshooting.
(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Patches:
works_on_my_machine.patch uploaded by xytis (license 6558)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In fax_detect_framehook() a null pointer reference can occur where a
voice frame is processed but no dsp is attached to the fax detection
structure. The code block that rejects frames that detection cannot
be processed on is checking for dsp but falls through when it should
instead return, as this change implements.
(closes issue ASTERISK-22942)
Reported by: adomjan
Review: https://reviewboard.asterisk.org/r/3076/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.
As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.
Much thanks to Torrey as well for providing a scenario that reproduces this
issue.
(closes issue ASTERISK-21242)
Reported by: Ashley Winters
Tested by: Torrey Searle
patches:
always-init-t38.patch uploaded by awinters (License 6477)
A_PARTY.xml uploaded by tsearle (License 5334)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.
(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r400089, a patch was put in to correct erroneous RTCP statistic resets.
Unfortunately, ast_rtp_read can be called on an RTP instance that does not
have RTCP information. This patch prevents that crash by only resetting
the statistics if we do actually have an RTCP instance.
(issue AST-1174)
(closes issue ASTERISK-22667)
Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.
(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.
(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.
(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RTCP's calculation of the number of lost packets in an RTP stream is based on
that stream's sequence number count, the number of received packets, and how
many packets we expect to receive. When the SSRC for an RTP stream changes,
there can - and almost always will be - a large jump in the next packet's
timestamp and sequence number. If we don't reset the number of received
packets, sequence number count, and other metrics used by RTCP, the next RR/SR
report will use the previous SSRC's values to calculate the lost packet count
for the new SSRC - resulting in a very large number of lost packets.
This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it
will reset the various values used by the RTCP calculations. From the
perspective of RTCP, this appears as a new media stream - which is what it is.
Review: https://reviewboard.asterisk.org/r/2886/
(closes issue AST-1174)
Reported by: Thomas Arimont
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sometimes the Google Voice servers have a bad habit of sending out 1
byte replies to the xmpp resource. When a blank 1 byte reply is
received from the socket the buffer attempts to wait (endlessly) for
the rest of the reply from google which effectively blocks the socket
and google voice calls will no longer come into the server.
This patch allows the xmpp module to correctly detect empty packets and
send out ping replies to google. It also sets a socket timeout on the
default socket which prevents the xmpp socket from closing and
preventing future google voice calls from coming into the server.
Furthermore instead of sending an empty reply back to google we send a
proper xmpp ping reply back. This also adds several more
socket messages.
(closes issue ASTERISK-22347)
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/2771
Patches:
xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The mailbox and context are swapped on the receiving end for all users
of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
versions. This swaps those values to be correct when publishing to the
internal event system from Jabber/XMPP distributed MWI state.
(closes issue ASTERISK-22435)
Reported by: abelbeck
Tested by: Michael Keuter
Patches:
asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_xmldoc_printable returns an allocated block that must be freed by the
caller. Fixed manager.c and res_agi.c to stop leaking these results.
(closes issue ASTERISK-22395)
Reported by: Corey Farrell
Patches:
manager-leaks-11.patch uploaded by coreyfarrell (license 5909)
res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If there is an error streaming an audio file, the current return status makes it
difficult for an AGI script to determine that there was an error with the audio
file.
This patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of
res_agi, this would appear to be the proper way to handle an error.
(closes issue ASTERISK-21903)
Reported by: Ariel Wainer
Tested by: Ariel Wainer
Patches:
asterisk-21903-return-stream-res_1.8.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2625/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes the following memory leaks:
* http.c: The structure containing the addresses to bind to was not being
deallocated when no longer used
* named_acl.c: The global configuration information was not disposed of
* config_options.c: An invalid read was occurring for certain option types.
* res_calendar.c: The loaded calendars on module unload were not being
properly disposed of.
* chan_motif.c: The format capabilities needed to be disposed of on module
unload. In addition, this now specifies the default options for the
maxpayloads and maxicecandidates in such a way that it doesn't cause the
invalid read in config_options.c to occur.
(issue ASTERISK-21906)
Reported by: John Hardin
patches:
http.patch uploaded by jhardin (license 6512)
named_acl.patch uploaded by jhardin (license 6512)
config_options.patch uploaded by jhardin (license 6512)
res_calendar.patch uploaded by jhardin (license 6512)
chan_motif.patch uploaded by jhardin (license 6512)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@392810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The WebSocket code would allocate, on the stack, a string large enough
to hold a key provided by the client, and the WEBSOCKET_GUID. If the key
is NULL, this causes a segfault. If the key is too large, it could
overflow the stack.
This patch checks the key for NULL and checks the length of the key to
avoid stack smashing nastiness.
(closes issue ASTERISK-21825)
Reported by: Alfred Farrugia
Tested by: Alfred Farrugia, David M. Lee
Patches:
issueA21825_check_if_key_is_sent.patch uploaded by Walter Doekes (license 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we send out a CN packet (for instance, in the case of using rtpkeepalives),
we are not setting the payload code properly. Also, we are setting the marker
bit when we shouldn't be according to RFC 3389, section 4.
AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using
ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup().
11 and trunk already use the appropriate function.
* In 1.8, use ast_rtp_codecs_payload_code()
* Remove the setting of the marker bit
* Fix the debug message by incrementing the seqno after the debug message is set
in order to display the correct seqno that was sent out
(closes issue ASTERISK-21246)
Reported by: Peter Katzmann
Tested by: Peter Katzmann, Michael L. Young
Patches:
asterisk-21246-rtp-cng-payload-error_1.8_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2500/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In certain situations, when the RTP engine goes to send a DTMF end digit
it may be in a situation where the remote address is no longer available,
or the digit that was supposed to be sent is invalid. In such cases, we
need to clear the RTP counters appropriately. Otherwise, when the RTP
source is set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party (signficantly).
(closes issue ASTERISK-21522)
Reported by: Corey Farrell
patches:
rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were several reports of deadlock when using
res_timing_pthread. Backtraces indicated that one thread was blocked
waiting for the write to the pipe to complete and this thread held
the container lock for the timers. Therefore any thread that wanted
to create a new timer or read an existing timer would block waiting
for either the timer lock or the container lock and deadlock ensued.
This patch changes the way the pipe is used to eliminate this source
of deadlocks:
1) The pipe is placed in non-blocking mode so that it would never
block even if the following changes someone fail...
2) Instead of writing bytes into the pipe for each "tick" that's
fired the pipe now has two states--signaled and unsignaled. If
signaled, the pipe is hot and any pollers of the read side
filedescriptor will be woken up. If unsigned the pipe is idle. This
eliminates even the chance of filling up the pipe and reduces the
potential overhead of calling unnecessary writes.
3) Since we're tracking the signaled / unsignaled state, we can
eliminate the exta poll system call for every firing because we know
that there is data to be read.
(closes issue ASTERISK-21389)
Reported by: Matt Jordan
Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis
patches:
0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417)
(closes issue ASTERISK-19754)
Reported by: Nikola Ciprich
(closes issue ASTERISK-20577)
Reported by: Kien Kennedy
(closes issue ASTERISK-17436)
Reported by: Henry Fernandes
(closes issue ASTERISK-17467)
Reported by: isrl
(closes issue ASTERISK-17458)
Reported by: isrl
Review: https://reviewboard.asterisk.org/r/2441/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_xmpp was not adding AST_EVENT_IE_CACHABLE to the event as each message came in,
then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event,
so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE.
(issue ASTERISK-20175)
(closes issue ASTERISK-21429)
(closes issue ASTERISK-21069)
(closes issue ASTERISK-21164)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2452/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the event as each message came in,
then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event,
so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE.
(issue ASTERISK-20175)
(closes issue ASTERISK-21429)
(closes issue ASTERISK-21069)
(closes issue ASTERISK-21164)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2452/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.
(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
rtp-timestamp.patch uploaded by pbertera (License 5943)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pjproject is what actually requires libuuid.
(closes issue ASTERISK-21125)
reported by Private Name
(Ed. note: Really? Private Name? I am rolling my eyes so hard right now.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk:
invalid pointer errors) can occur as the memory is being allocated with
Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the
LDAP library's wrappers.
This patch uses the LDAP library's wrappers where appropriate, so that
compiling with MALLOC_DEBUG doesn't cause more problems than it solves.
Note that the patch listed below was modified slightly for this commit
to account for some additional memory allocation/deallocations.
(closes issue ASTERISK-17386)
Reported by: John Covert
Tested by: Andrew Latham
patches:
issue18789-1.8-r316873.patch uploaded by seanbright (License 5060)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both endpoints
depending on the call legs involved). The test now properly checks the
version field in the RTP header to ensure that RTP and RTCP are
decrypted while other types of packets are not.
(closes issue ASTERISK-21323)
Reported by: andrea
Tested by: Kinsey Moore, andrea, John Bigelow
Patches:
whitenoise_fix.diff uploaded by Kinsey Moore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The format attribute resource for H.264 video performs an unsafe read against a
media attribute when parsing the SDP. The value passed in with the format
attribute is not checked for its length when parsed into a fixed length buffer.
This patch resolves the vulnerability by only reading as many characters from
the SDP value as will fit into the buffer.
(closes issue ASTERISK-20901)
Reported by: Ulf Harnhammar
patches:
h264_overflow_security_patch.diff uploaded by jrose (License 6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
