If to_answer is -1, simply comparing to see if the progress timeout
is smaller than the answer timeout to prefer it will fail. Add
an additional check that chooses the progress timeout if there is
no answer timeout (or as before, if the progress timeout is smaller).
Resolves: #821
* pjproject is now configured with --disable-libsrtp so it will
build correctly when doing "out-of-tree" development. Asterisk
doesn't use pjproject for handling media so pjproject doesn't
need libsrtp itself.
* The pjsua app (which we used to use for the testsuite) no longer
builds in pjproject's master branch so we just skip it. The
testsuite no longer needs it anyway.
See third-party/pjproject/README-hacking.md for more info on building
pjproject "out-of-tree".
This reverts commit cb5e3445be.
The original change from 16 to 15 bit sequence numbers was predicated
on the following from the now-defunct libSRTP FAQ on sourceforge.net:
> *Q6. The use of implicit synchronization via ROC seems
> dangerous. Can senders and receivers lose ROC synchronization?*
>
> **A.** It is possible to lose ROC synchronization between sender and
> receiver(s), though it is not likely in practice, and practical
> steps can be taken to avoid it. A burst loss of 2^16 packets or more
> will always break synchronization. For example, a conversational
> voice codec that sends 50 packets per second will have its ROC
> increment about every 22 minutes. A network with a burst of packet
> loss that long has problems other than ROC synchronization.
>
> There is a higher sensitivity to loss at the very outset of an SRTP
> stream. If the sender's initial sequence number is close to the
> maximum value of 2^16-1, and all packets are lost from the initial
> packet until the sequence number cycles back to zero, the sender
> will increment its ROC, but the receiver will not. The receiver
> cannot determine that the initial packets were lost and that
> sequence-number rollover has occurred. In this case, the receiver's
> ROC would be zero whereas the sender's ROC would be one, while their
> sequence numbers would be so close that the ROC-guessing algorithm
> could not detect this fact.
>
> There is a simple solution to this problem: the SRTP sender should
> randomly select an initial sequence number that is always less than
> 2^15. This ensures correct SRTP operation so long as fewer than 2^15
> initial packets are lost in succession, which is within the maximum
> tolerance of SRTP packet-index determination (see Appendix A and
> page 14, first paragraph of RFC 3711). An SRTP receiver should
> carefully implement the index-guessing algorithm. A naive
> implementation can unintentionally guess the value of
> 0xffffffffffffLL whenever the SEQ in the packet is greater than 2^15
> and the locally stored SEQ and ROC are zero. (This can happen when
> the implementation fails to treat those zero values as a special
> case.)
>
> When ROC synchronization is lost, the receiver will not be able to
> properly process the packets. If anti-replay protection is turned
> on, then the desynchronization will appear as a burst of replay
> check failures. Otherwise, if authentication is being checked, then
> it will appear as a burst of authentication failures. Otherwise, if
> encryption is being used, the desynchronization may not be detected
> by the SRTP layer, and the packets may be improperly decrypted.
However, modern libSRTP (as of 1.0.1[1]) now mentions the following in
their README.md[2]:
> The sequence number in the rtp packet is used as the low 16 bits of
> the sender's local packet index. Note that RTP will start its
> sequence number in a random place, and the SRTP layer just jumps
> forward to that number at its first invocation. An earlier version
> of this library used initial sequence numbers that are less than
> 32,768; this trick is no longer required as the
> rdbx_estimate_index(...) function has been made smarter.
So truncating our initial sequence number to 15 bit is no longer
necessary.
1. 0eb007f0dc/CHANGES (L271-L289)
2. 2de20dd9e9/README.md (implementation-notes)
When the channel tech is multistream capable, the reference to
chan_topology was passed to the new channel. When the channel tech
isn't multistream capable, the reference to chan_topology was never
released. "Local" channels are multistream capable so it didn't
affect them but the confbridge "CBAnn" and the bridge_media
"Recorder" channels are not so they caused a leak every time one
of them was created.
Also added tracing to ast_stream_topology_alloc() and
stream_topology_destroy() to assist with debugging.
Resolves: #938
The dnsmgr_refresh() function checks to see if the IP address associated
with a name/service has changed. The gotcha is that the ast_get_ip_or_srv()
function only returns the first IP address returned by the DNS query. If
there are multiple IPs associated with the name and the returned order is
not consistent (e.g. with DNS round-robin) then the other IP addresses are
not included in the comparison and the entry is flagged as changed even
though the IP is still valid.
Updated the code to check all IP addresses and flag a change only if the
original IP is no longer valid.
Resolves: #924
Under some circumstances, the progress timeout feature added in commit
320c98eec8 does not work as expected,
such as if there is no media flowing. Adjust the waitfor call to
explicitly use the progress timeout if it would be reached sooner than
the answer timeout to ensure we handle the timers properly.
Resolves: #821
In some circumstances, it is possible for the do_monitor thread to
erroneously think that a line is on-hook and send an MWI FSK spill
to it when the line is really off-hook and no MWI should be sent.
Commit 0a8b3d3467 previously fixed this
issue in a more readily encountered scenario, but it has still been
possible for MWI to be sent when it shouldn't be. To robustly fix
this issue, query DAHDI for the hook status to ensure we don't send
MWI on a line that is actually still off hook.
Resolves: #928
Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be
balanced to ensure that we can capture the exit status of child
processes when we need to. This extends to functions that call
`ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as
`ast_safe_fork()` and `ast_safe_fork_cleanup()`.
The primary change here is ensuring that we do not call
`ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously
called `ast_safe_fork()`.
Additionally we reinforce some of the documentation and add an
assertion to, ideally, catch this sooner were this to happen again.
Fixes#922
asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings.
config.c: Include header file for WIFEXITED/WEXITSTATUS macros.
res_timing_kqueue: Use more portable format specifier.
test_crypto: Use non-linux limits.h header file.
Resolves: #916
In dtls_srtp_handle_timeout(), when DTLSv1_get_timeout() returned
success but with a timeout of 0, we were stopping the timer and
decrementing the refcount on instance but not resetting the
timeout_timer to -1. When dtls_srtp_stop_timeout_timer()
was later called, it was atempting to stop a stale timer and could
decrement the refcount on instance again which would then cause
the instance destructor to run early. This would result in either
a FRACK or a SEGV when ast_rtp_stop(0 was called.
According to the OpenSSL docs, we shouldn't have been stopping the
timer when DTLSv1_get_timeout() returned success and the new timeout
was 0 anyway. We should have been calling DTLSv1_handle_timeout()
again immediately so we now reschedule the timer callback for
1ms (almost immediately).
Additionally, instead of scheduling the timer callback at a fixed
interval returned by the initial call to DTLSv1_get_timeout()
(usually 999 ms), we now reschedule the next callback based on
the last call to DTLSv1_get_timeout().
Resolves: #487
When using the ModuleLoad AMI action, it was possible to traverse
upwards through the directories to files outside of the configured
modules directory. We decided it would be best to restrict access to
modules exclusively in the configured directory. You will now get an
error when the specified module is outside of this limitation.
Fixes: #897
UserNote: The ModuleLoad AMI action now restricts modules to the
configured modules directory.
When using the speech recognition module, crashes can occur
sporadically due to a "double free or corruption (out)" error. Now, in
the section where the audio stream is being captured in a loop, each
time after releasing fr, it is set to NULL to prevent repeated
deallocation.
Fixes#772
Don't pass through a NULL argument to fclose, which is undefined
behavior, and instead return -1 and set errno appropriately. This
also avoids a compiler warning with glibc 2.38 and newer, as glibc
commit 71d9e0fe766a3c22a730995b9d024960970670af
added the nonnull attribute to this argument.
Resolves: #900
In certain circumstances a channel may undergo an operation
referred to as a masquerade. If this occurs the CHANNEL(userfield)
value was not preserved causing it to get lost. This change makes
it so that this field is now preserved.
Fixes: #882
attest_level, send_mky and check_tn_cert_public_url weren't
propagating correctly from the attestation object to the profile
and tn.
* In the case of attest_level, the enum needed to be changed
so the "0" value (the default) was "NOT_SET" instead of "A". This
now allows the merging of the attestation object, profile and tn
to detect when a value isn't set and use the higher level value.
* For send_mky and check_tn_cert_public_url, the tn default was
forced to "NO" which always overrode the profile and attestation
objects. Their defaults are now "NOT_SET" so the propagation
happens correctly.
* Just to remove some redundant code in tn_config.c, a bunch of calls to
generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were
replaced with a single call to generate_acfg_common_sorcery_handlers().
Resolves: #904
The app is actually named "BackGround" but several references
in XML documentation were spelled "Background" with the lower
case "g". This was causing documentation links to return
"not found" messages.
UserNote: You can now perform more granular filtering on events
in manager.conf using expressions like
`eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
This is much more efficient than
`eventfilter = Event: Newchannel.*Channel: PJSIP/`
Full syntax guide is in configs/samples/manager.conf.sample.
Consumers like media_cache have been running into issues with
the previous astdb "/family/key" limit of 253 bytes when needing
to store things like long URIs. An Amazon S3 URI is a good example
of this. Now, instead of using a static 256 byte buffer for
"/family/key", we use ast_asprintf() to dynamically create it.
Both test_db.c and test_media_cache.c were also updated to use
keys/URIs over the old 253 character limit.
Resolves: #881
UserNote: The `ast_db_*()` APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the `DB*` dialplan applications, dialplan functions,
manager actions and `databse` CLI commands. Since the
media_cache also uses the `ast_db_*()` APIs, you can now store
resources with URIs longer than 253 bytes.
verification.c had an include for jansson.h left over from previous
versions of the module. Since res_stir_shaken no longer has a
dependency on jansson, the bundled version wasn't added to GCC's
include path so if you didn't also have a jansson development package
installed, the compile would fail. Removing the stale include
was the only thing needed.
Resolves: #889
* If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID,
check_for_old_config() now returns LOAD_DECLINE instead of continuing
on with a bad pointer.
* If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config()
assumes the config is being loaded from realtime and now returns
LOAD_SUCCESS. If it's actually not being loaded from realtime,
sorcery will catch that later on.
* Also refactored the error handling in load_module() a bit.
Resolves: #884
For both attestation and verification, we now check whether they've
been disabled either globally or by the profile before validating
things like callerid, orig_tn, dest_tn, etc. This prevents useless
error messages.
Resolves: #879
ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.
Resolves: #876
It's possible that ast_autoservice_stop is called within the autoservice thread.
In this case the autoservice thread is stuck in an endless sleep.
To avoid endless sleep ast_autoservice_stop must check that it's not called
within the autoservice thread.
Fixes: #763
The ub_result pointer passed to unbound_resolver_callback by
libunbound can be NULL if the query was for something malformed
like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result
and return instead of crashing with a SEGV. This causes pjproject
to simply cancel the transaction with a "No answer record in the DNS
response" error. The existing "off nominal" unit test was also
updated to check this condition.
Although not necessary for this fix, we also made
ast_dns_resolver_completed() tolerant of a NULL result.
Resolves: GHSA-v428-g3cw-7hv9
When commit e8c9cb80 was cherry-picked in from master, the
fact that the 20 and 18 branches still had the old "macrocontext"
column wasn't taken into account so the number of named parameters
didn't match the number of '?' placeholders. They do now.
We also now use ast_asprintf to create the full mailbox query SQL
statement instead of trying to calculate the proper length ourselves.
Resolves: #831
When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
matching 4733 offers, Bob may want to use the 48K audio codec but can not
accept 48K digits and so negotiates for a mixed set.
Asterisk will now check Bob's offer to make sure Bob has indicated this is
acceptible and if not, will use Bob's preference.
Fixes: #847
ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail.
Resolves: #861
* A static array of security mechanism type names was created.
* ast_sip_str_to_security_mechanism_type() was refactored to do
a lookup in the new array instead of using fixed "if/else if"
statments.
* security_mechanism_to_str() and ast_sip_security_mechanisms_to_str()
were refactored to use ast_str instead of a fixed length buffer
to store the result.
* ast_sip_security_mechanism_type_to_str was removed in favor of
just referencing the new type name array. Despite starting with
"ast_sip_", it was a static function so removing it doesn't affect
ABI.
* Speaking of "ast_sip_", several other static functions that
started with "ast_sip_" were renamed to avoid confusion about
their public availability.
* A few VECTOR free loops were replaced with AST_VECTOR_RESET().
* Fixed a meomry leak in pjsip_configuration.c endpoint_destructor
caused by not calling ast_sip_security_mechanisms_vector_destroy().
* Fixed a memory leak in res_pjsip_outbound_registration.c
add_security_headers() caused by not specifying OBJ_NODATA in
an ao2_callback.
* Fixed a few ao2_callback return code misuses.
Resolves: #845
PR #700 added a preferred_format for the struct ast_rtp_codecs,
but when set the preferred_format it leaks an astobj2 ast_format.
In the next code
ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0));
both functions ast_rtp_codecs_set_preferred_format
and ast_format_cap_get_format increases the ao2 reference count.
Fixes: #856
On modern Bluetooth devices or lower-powered asterisk servers, decreasing the channel frame size significantly improves latency and delay on outbound calls with only a mild sacrifice to the quality of the call (the frame size before was massive overkill to begin with)
Add dialplan application PJSIPNOTIFY to send either pre-configured
NOTIFY messages from pjsip_notify.conf or with headers defined in
dialplan.
Also adds the ability to send pre-configured NOTIFY commands to a
channel via the CLI.
Resolves: #799
UserNote: A new dialplan application PJSIPNotify is now available
which can send SIP NOTIFY requests from the dialplan.
The pjsip send notify CLI command has also been enhanced to allow
sending NOTIFY messages to a specific channel. Syntax:
pjsip send notify <option> channel <channel>
If you run an AMI CoreShowChannelMap on a channel that isn't in a
bridge and you're in DEVMODE, you can get a FRACK because the
bridge id is empty. We now simply return an empty list for that
request.
