Use the new ast_cli_completion_add() function to improve completion
performance for commands like 'pjsip show endpoint.'
Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348
Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.
This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.
Related to ASTERISK~26806, but does not completely resolve it.
Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
res_stasis was missing AST_MODFLAG_LOAD_ORDER. Set res_stasis and
res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for
dependent modules.
Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3
For both dynamic and static contacts it was possible that potential AOR
changes were not being applied to all contacts. This was because the qualify
and schedule code was only retrieving AOR's, and contacts with frequencies
greater than zero.
For instance the following could happen: and AOR/contact has a frequency of 5,
it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are
stopped, a list of AOR's is retrieved with frequency > 0, but none are
selected since in this scenario all are 0. The contact for the one previously
set to 5 though does not get updated, so it's status remains "AVAILABLE".
This patch makes it so all contacts (static and dynamic) are selected, and
appropriately updated if need be.
ASTERISK-27467 #close
Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb
Dynamic contacts were not being properly updated on reload. As a matter of
fact any changes to the AOR that a dynamic contact was associated with were
not being applied.
On reload, this patch makes it so for each dynamic contact, the associated
AOR is now retrieved and the AOR's fields are applied to the contact.
ASTERISK-27467
Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
More complicated direct media reinvite negotiations can result in longer
delays before direct media flows. The strictrtp learning timeout time
was too short. One log showed that the first RTP packet came in just
after three seconds.
* Increase the strictrtp learning timeout time from 1.5 to 5 seconds.
ASTERISK-27453
Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c
This change makes the presence of the GMIME_MAJOR_VERSION
definition optional, as not all versions of gmime actually
define it.
ASTERISK-27454
Change-Id: I01d99590045971ed6787899147170a5954077238
The patch for ASTERISK_24560 inverted a test checking if the bridge name
is being updated to a different name.
* Fix the test to return "Changing bridge name is not implemented" when
someone attempts to change the bridge name.
ASTERISK-27445
Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9
Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now,
srflx is gathered and sent via SDP, even when Asterisk is enabled for
Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used.
ASTERISK-27437
Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1
res_parking had an inplicit load_pri of 0 meaning it was one of the very
first modules loaded after modules with global symbols. Set it to
AST_MODPRI_DEVSTATE_PROVIDER as it provides device state for parking
lots.
Change-Id: I297b6fb3ff6993ec004e667b22a74f5925906259
res_mwi_external_ami specified AST_MODFLAG_LOAD_ORDER but didn't set
load_pri, resulting in an actual load priority of 0. This module only
provides AMI actions so it has no reason to load early.
Change-Id: I82987fcf10d3ea42716b2f9df915b16687fd5839
Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri
AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT.
Change-Id: I0123258eafce324249433a69df15a85cc16e509f
Mac doesn't like the comparison of -1 to an enum, so store the result of
ast_sip_str_to_dtmf to an int so we can check for the negative return
value. ast_sip_str_to_dtmf returns an int so this is only delaying the
implicit type cast.
Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29
Some net-snmp builds do not provide the RONLY declare, only
NETSNMP_OLDAPI_RONLY. Map RONLY to NETSNMP_OLDAPI_RONLY to get around
this error.
Change-Id: Ida5c7ad9406515825485c4d3b4a34fd6ad0da577
It's impossible for gwtimeout or fdtimeout to be less than 0 because
they are unsigned int's. Remove checks and unreachable branches.
Change-Id: Ib2286960621e6ee245e40013c84986143302bc78
Some clients do not send rtp packets every ptime ms. This can lead to
situations in which the rtp source learning algorithm will never learn
the address of the client. This has been discovered on a Mac mini with
a pjsip based softphone after updating to Sierra: as soon as USB
headsets are involved, the softphone will send the second packet 30ms
after the first, the third 30ms after the second and the fourth 1ms
after the third. So in the old implmentation the rtp source learning
algorithm was repeatedly reset on the fourth packet.
The patch changes the algorithm in a way that doesn't take the arrival
time between two consecutive packets into account but the time between
the first and the last packet of a learning sequence.
The patch also fixes a second problem: when a user was using a wrong
value for the probation setting there was a LOG_WARNING output stating
that the value had been set to the default value instead. However
the code for setting the value back to defaults was missing.
ASTERISK-27421 #close
Change-Id: If778fe07678a6fd2041eaca7cd78267d0ef4fc6c
Domains themselves can be up to 255 characters long (per RFC 1035), so
our current buffer sizes are wholly inadequate for many use cases.
Change-Id: If3f30a68307f1365a1fe06bc4b854c62842c9292
We were not \0 terminating this string, so any attempt to print it would
in the best case show an empty string and in the worst case potentially
crash.
Change-Id: I63d96ef8f7516ac02a0f91e22dfa8acdc615042c
Previously for PJSIP the local address of WebSocket connections
was set to the remote address. For logging purposes this is
not particularly useful.
The WebSocket API has been extended to allow the local
address to be queried and this is used in PJSIP to set the
local address to the correct value.
The PJSIP HEP support has also been tweaked so that reliable
transports always use the local address on the transport
and do not try to (wrongly) guess. As they are connection
based it is impossible for the source to be anything else.
ASTERISK-26758
ASTERISK-27363
Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca
Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
A previous commit made it so when an invite session transitioned into a
disconnected state destruction of the Asterisk pjsip session object was
postponed until either a transport error occurred or the event timer
expired. However, if a call was rejected (for instance a 488) before the
session was fully established the event timer may not have been initiated,
or it was canceled without triggering either of the session finalizing states
mentioned above.
Really the only time destruction of the session should be delayed is when a
BYE is being transacted. This is because it's possible in some cases for the
session to be disconnected, but the BYE is still transacting.
This patch makes it so the session object always gets released (no more
memory leak) when the pjsip session is in a disconnected state. Except when
the method is a BYE. Then it waits until a transport error occurs or an event
timeout.
ASTERISK-27345 #close
Reported by: Corey Farrell
Change-Id: I1e724737b758c20ac76d19d3611e3d2876ae10ed
* Pre-initialize cloned media state vectors to final size to ensure
vector errors cannot happen later in the clone initialization.
* Release session_media on vector replace failure in
ast_sip_session_media_state_add.
* Release clone and media_state in ast_sip_session_refresh if we fail to
append to the stream topology, return an error.
Change-Id: Ib5ffc9b198683fa7e9bf166d74d30c1334c23acb
One of the patches for ASTERISK_27147 introduced a deadlock regression.
When the connection oriented transport shut down, the code attempted to
remove the associated contact. However, that same transport had just
requested a registration that we hadn't responded to yet. Depending
upon timing we could deadlock.
* Made send the REGISTER response after we completed processing the
request contacts and released the AOR lock to avoid the deadlock.
ASTERISK-27391
Change-Id: I89a90f87cb7a02facbafb44c75d8845f93417364
* res/stasis/app.c JSON passed to app_send needs to be released.
* res/stasis_message.c: objects leak if vector append fails.
Change-Id: I8dd5385b9f50a5cadf2b1d16efecffd6ddb4db4a
Asterisk will crash if contact uri is invalid, so contact_apply_handler
should check if the uri is NULL or empty.
ASTERISK-27393 #close
Reported-by: Aaron An
Tested-by: AaronAn
Change-Id: Ia0309bdc6b697c73c9c736e1caec910b77ca69f5
wizard_apply_handler():
- Free host if we fail to add it to the vector.
wizard_mapped_observer():
- Check for otw allocation failure.
- Free otw if we fail to add it to the vector.
Change-Id: Ib5d3bcabbd9c24dd8a3c9cc692a794a5f60243ad
When stasis_app_message_handler needs to queue a message for a later
connection it needs to bump the message reference so it doesn't get
freed when the caller releases it's reference.
Change-Id: I82696df8fe723b3365c15c3f7089501da8daa892
This change makes it so that any user of the pubsub
API that requests the remote URI receives only the URI.
Previously the entire string was returned, which could
contain a display name.
ASTERISK-27290
Change-Id: If1d0cd6630f0a264856d31d2a67933109187a017
When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it. The most likely way it can return failure is for an
allocation failure. If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.
* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure. That is much more useful than either an uninitialized pointer
or a pointer that has already been freed. Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.
* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().
* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.
Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23
When using realtime, fields that are not explicitly set by an
administrator are still presented to sorcery as empty strings. Handle
this case explicitly.
In this particular case, if any of these fields are required for TLS
support, their existence should be validated in the 'apply' handler once
we have a complete transport definition.
ASTERISK-27032 #close
Reported by: seanchann.zhou
Change-Id: Ie3b5fb421977ccdb33e415d4ec52c3fd192601b7
This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.
Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.
ASTERISK-27395
Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
Fixes a regression where some characters were unable to be used in
the from_user field of an endpoint. Additionally, the backtick was
removed from the list of valid characters, since it is not valid,
and it was replaced with a single quote, which is a valid character.
ASTERISK-27387
Change-Id: Id80c10a644508365c87b3182e99ea49da11b0281
Once an Optional API module is loaded it should stay loaded. Unloading
an optional API module runs the risk of a crash if something else is
using it. This patch causes all optional API providers to tell the
module loader not to unload except at shutdown.
ASTERISK-27389
Change-Id: Ia07786fe655681aec49cc8d3d96e06483b11f5e6
channel_state_invalid leaked a reference to the channel snapshot any
time it was aquired.
ASTERISK-27067 #close
Change-Id: I8c653f00416b39978513c5605c4be0f03b1df29a
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.
This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.
ASTERISK-27379
Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
When allocate_subscription fails to initialize fields of the new sub it
calls destroy_subscription.
Change-Id: I5b79c915ec216dc00c13c1e4172137864a4bec85
When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.
ASTERISK-27206
Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd
create_outgoing_sdp_stream was setting "addr_type = STR_IP6" only
when an ipv6 media_address was specified on the endpoint. If
rtp_ipv6 was set and ast_sip_get_host_ip_string returned an ipv6
address, we were leaving the addr_type set at the default of
STR_IP4. This caused the address type to be set incorrectly on the
"o" and "c" SDP attributes even though the address was set
correctly. Some clients don't like the mismatch.
* Removed the test for endpoint/media_address and now check all
addresses for ipv6.
ASTERISK-27198
Reported by: Martin Cisárik
Change-Id: I5214fc31b728117842243807e7927a319cf77592
Users of the API that res_xmpp provides expect that a
filter be available on the client at all times. When
OAuth authentication support was added this requirement
was not maintained.
This change merely moves the OAuth authentication to
after the filter is created, ensuring users of res_xmpp
can add things to the filter as needed.
ASTERISK-27346
Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886
Prevent unload of the module as certain pjsip initialization functions
cannot be reversed. This required a reorder of the module_load so that
the non-reversable pjsip functions are not called until all potential
errors have been ruled out.
ASTERISK-24483
Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7
Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.
ASTERISK-24483
Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d
PJSIP allows a domain name as external_media_address. This allows chan_pjsip to
be used behind a NAT with changing IP addresses. The IP address of that domain
is resolved to the c= line already. This change sets also the o= line to that
domain.
ASTERISK-27341 #close
Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4
Move ast_sip_add_usereqphone to be called after anonymization of URIs,
to prevent the user_eq_phone adding "user=phone" to URIs containing a
username that is not a phonenumber (RFC3261 19.1.1). An extra call to
ast_sip_add_usereqphone on the saved version before anonymization is
added to add user=phone" to the PAI.
ASTERISK-27047 #close
Change-Id: Ie5644bc66341b86dc08b1f7442210de2e6acdec6
ast_sip_add_usereqphone adds "user=phone" to the header every time is is
called without checking whether the param already exists. Preventing
this by searching to string representation of header for "user=phone".
ASTERISK-26988 #close
Change-Id: Ib84383b07254de357dc6a98d91fc1d2c2c3719e6
Add bridge_features structure to bridge creation. Specifically, this
implements mute and DTMF suppression, but others should be able to be
easily added to the same structure.
ASTERISK-27322 #close
Reported by: Darren Sessions
Sponsored by: AVOXI
Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61
When "rewrite_contact" is enabled, the "max_contacts" count option can
block re-registrations because the source port from the endpoint can be
random. When the re-registration is blocked, the endpoint may give up
re-registering and require manual intervention.
* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire. The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one. The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.
ASTERISK-27192
Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b
Do not manually call sip_endpoint_apply_handler from load_all_endpoints.
This is not necessary and causes memory leaks.
Additionally reinitialize persistent->aors when we reuse a persistent
object with a new endpoint.
ASTERISK-27306
Change-Id: I59bbfc8da8a14d5f4af8c5bb1e71f8592ae823eb
pjsip_distributor leaks references to fake_auth when the default realm
has not changed.
ASTERISK-27306
Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202
Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).
ASTERISK-27284 #close
Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56
This provides better information to REF_DEBUG log for troubleshooting
when the system is unable to unload res_pjsip.so during shutdown due to
module references.
ASTERISK-27306
Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612
res_pjsip and res_pjsip_session had circular references, preventing both
modules from shutting down.
* Move session supplement registration to res_pjsip.
* Use create internal functions for use by pjsip_message_filter.c.
ASTERISK-27306
Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b
When we are loading the calendars, we call libical's
icalcomponent_foreach_recurrence method for each VEVENT component that
we have in our calendar.
That method has no knowledge concerning the existence of the other
VEVENT components and will feed our callback with all ocurrences
matching the requested time span.
The occurrences generated by icalcomponent_foreach_recurrence while
expanding a recurring VEVENT's RRULE and RDATE properties can be
superceded by an other VEVENT sharing the same UID.
I use an external iterator (in libical terminology) to avoid messing
with the internal ones from the calling function, and search for
VEVENTS which could supersede the current occurrence.
The event which can invalidate this occurence needs to have:
- the same UID as our recurrent component (comp)
- a RECURRENCE-ID property, which represents the start time of this
occurrence
If one component is found, just clean and return.
ASTERISK-27296 #close
Reported by: Benoît Dereck-Tricot
Change-Id: I8587ae3eaa765af7cb21eda3b6bf84e8a1c87af8
This change adds 'video_sfu' as a requested bridge type when
creating a bridge. By specifying this a mixing type bridge is
created that exchanges video in an SFU fashion.
Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606
The pjsip_publishc_init() call was referenced with a misplaced
parentheses. As a result, outbound publication messages went out with an
expiration of 1 second.
ASTERISK-27298
Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08
The "res_pjsip: Filter out non SIP(S) requests" commit moved the
filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER
in order to filter out incoming bad uri schemes as early as possible.
Since the change affected outgoing messages as well and the TRANSPORT
layer is the last to be run on outgoing messages, we were overwriting
the setting of external_signaling_address (which is set earlier by
res_pjsip_nat) with an internal address.
* pjsip_message_filter now registers itself as a pjproject module
twice. Once in the TSX layer for the outgoing messages (as it was
originally), then a second time in the TRANSPORT layer for the
incoming messages to catch the invalid uri schemes.
ASTERISK-27295
Reported by: Sean Bright
Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c
The bridge_p2p_rtp_write() has potential reentrancy problems.
* Accessing the bridged RTP members must be done with the instance1 lock
held. The DTMF and asymmetric codec checks must be split to be done with
the correct RTP instance struct locked. i.e., They must be done when
working on the appropriate side of the point to point bridge.
* Forcing the RTP mark bit was referencing the wrong side of the point to
point bridge. The set mark bit is used everywhere else to set the mark
bit when sending not receiving.
The patches for ASTERISK_26745 and ASTERISK_27158 did not take into
account that not everything carried by RTP uses a codec. The telephony
DTMF events are not exchanged with a codec. As a result when
RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is
enabled, the DTMF digits would always get passed to the core even though
the local native RTP bridge is active, and the DTMF digits would go out
using the wrong SSRC id.
* Add protection for non-format payload types like DTMF when updating the
lastrxformat and lasttxformat. Also protect against non-format payload
types when checking for asymmetric codecs.
ASTERISK-27292
Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186
This could have been fixed by subtracting 1 from the final value of
'len' but the way the packet was being constructed was confusing so I
took the opportunity to (I think) make it more clear.
We were sending 1 extra byte at the end of the SDES RTCP packet which
caused Chrome to complain (in its debug log):
Too little data (1 byte) remaining in buffer to parse
RTCP header (4 bytes).
We now send the correct number of bytes.
Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b
If using a legitimate certificate from a trusted certificate authority,
you don't need to provide CA file.
Change-Id: I8623973b4209b44889243716d7880274caed8a6d
For some scenarios when an outgoing call was made only a subset of the
configured codecs were offered. If the codecs being offered happened to
not have a codec supported by the phone then the call would fail.
For instance Alice and Bob both are configured in Asterisk for g722 and ulaw(
allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's
only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with
Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains
g722 and not both g722 and ulaw, so the call ends.
This patch makes it so all the audio codecs configured on the endpoint always
get sent, and not just a subset. However priority is given to those codecs that
are compatible with the "other side".
ASTERISK-27259 #close
Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
When pruning a request to change the topology of a channel be
more intelligent about the resulting topology that is actually
used for SDP renegotiation.
In a case where a stream has not already been negotiated we
don't need to renegotiate and offer a declined stream. This can
occur if something in Asterisk (such as ConfBridge) requests
to add video to a PJSIP channel that has no video codecs configured.
In this case since the stream did not already exist we can safely
remove the stream from the requested topology, resulting in no
renegotiation occurring.
In a case where a renegotiation is requested with a codec that is
not supported we can reuse the formats of the existing stream if
it exists to ensure that the stream continues to flow, instead of
removing it.
Change-Id: I636540798d55922377318fe619c510fb6ed125fb
During a reinvite, if a remote endpoint error occurs and it returns a 500 the
session would end. This patch makes it so the session is not terminated, but
continues as it was.
The reason for this is because some endpoints may send non session terminating
"server errors" like a failed codec negotiation. So in this case instead of
ending the call it can hopefully continue. In the case of a real server error
the session is already "doomed", will be known soon enough and appropriately
ended by Asterisk later.
Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5
