The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.
Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.
The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.
Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.
Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.
If a stream has been removed or declined we will now mark it as such
within the resulting SDP.
Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.
Two new configuration options have also been added to PJSIP endpoints:
max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.
max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.
ASTERISK-27076
Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.
ASTERISK-26919 #close
Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.
ASTERISK-26281
Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c
This option was added to turn off notifying the progress details
on Blind Transfer. If this option is not set then the chan_pjsip
will send NOTIFY "200 OK" immediately after "202 Accepted".
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
ASTERISK-26333 #close
Change-Id: Id606fbff2e02e967c02138457badc399144720f2
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
We were leaking a transport ref in multihomed_on_rx_message() which
resulted in the FRACK about excessive ref counts.
ASTERISK-26916 #close
Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f
Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.
ASTERISK-26864
Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6
A new transport parameter 'symmetric_transport' has been added.
When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output. On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.
* config_transport was modified to accept and store the new parameter.
* config_transport/transport_apply was updated to store the transport
name in the pjsip_transport->info field using the pjsip_transport->pool
on UDP transports.
* A 'multihomed_on_rx_message' function was added to
pjsip_message_ip_updater that, for incoming requests, retrieves the
transport name from pjsip_transport->info and retrieves the transport.
If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
containing the transport name is added to the incoming Contact header.
* An 'ast_sip_get_transport_name' function was added to res_pjsip.
It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
transport name if endpoint->transport is set or if there's an
'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
ipv6 address. Otherwise it returns NULL.
* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
pjsip_tpselector. It calls ast_sip_get_transport_name() and if
a non-NULL is returned, sets the selector and sets the transport
on the dialog. If a selector was passed in, it's updated.
* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
were modified to call ast_sip_dlg_set_transport() instead of their
original logic.
* res_pjsip/create_out_of_dialog_request was modified to call
ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
instead of its original logic.
* Existing transport logic was removed from endpt_send_request
since that can only be called after a create_out_of_dialog_request.
* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
a new 'ast_sip_create_rdata_with_contact' function which allows
a contact_uri to be specified in addition to the existing
parameters. (See below)
* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
since all it did was transport selection and that is now done in
ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.
* 'contact_uri' was added to subscription_persistence. This was
necessary because although the parsed rdata contact header has the
x-ast-txp parameter added (if appropriate),
subscription_persistence_update stores the raw packet which
doesn't have it. subscription_persistence_recreate was then
updated to call ast_sip_create_rdata_with_contact with the
persisted contact_uri so the recreated subscription has the
correct transport info to send the NOTIFYs.
* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
all it did was transport selection and that is now done in
ast_sip_create_dialog_uac.
* pjsip_message_ip_updater/multihomed_on_tx_message was updated
to remove all traces of the x-ast-txp parameter from the
outgoing headers.
NOTE: This change does NOT modify the behavior of permanent
contacts specified on an aor. To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated. If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.
You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.
Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f
This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.
A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.
The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.
ASTERISK-26732 #close
Reported by Dan Jenkins
Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.
* Use WSS in Via for secure transport.
* Only register one transport with the WS name because it would be
ambiguous. Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered. This may mess up unsecure
websockets but the impact should be minimal. Firefox and Chrome do not
support anything other than secure websockets anymore.
* Added and updated some debug messages concerning websockets.
* security_events.c: Relax case restriction when determining security
transport type.
* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.
[1] https://tools.ietf.org/html/rfc7118
ASTERISK-26796 #close
Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12
This change fixes an assumption in res_pjsip that a contact will
always have a status. There is a race condition where this is
not true and would crash. The status will now be unknown when
this situation occurs.
ASTERISK-26623 #close
Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5
There was code attempting to update the artificial authentication object
whenever the default_realm changed. However, once the artificial
authentication object was created it would never get updated. The
artificial authentication object would require a system restart for a
change to the default_realm to take effect.
ASTERISK-26799
Change-Id: Id59036e9529c2d3ed728af2ed904dc36e7094802
* Removed overloaded unmatched response ignore. We obviously sent the
request so we shouldn't ignore it because it isn't new work.
ASTERISK-26669
ASTERISK-26738
Change-Id: I55fb5cadc83a8e6699b347c6dc7fa32c5a617d37
When listing a container, we now print the number of objects
in the container at the end of the list.
Change-Id: I791cbc3ee9da9a2af9adc655164b5d32953df812
This patch fixes 2 original issues and more that those 2 exposed.
* When we send a NOTIFY, and the client either doesn't respond or
responds with a non OK, pjproject only calls our
pubsub_on_evsub_state callback, no others. Since
pubsub_on_evsub_state (which does the sub_tree cleanup) does not
expect to be called back without the other callbacks being called
first, it just returns leaving the sub_tree orphaned. Now
pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
which is what pjproject will set to tell us that it was the
transaction that timed out or failed and not the subscription
itself timing our or being terminated by the client. If is
TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
regardless of the state of the subscription.
* When a client renews a subscription, we don't update the
persisted subscription with the new expires timestamp. This causes
subscription_persistence_recreate to prune the subscription if/when
asterisk restarts. Now, pubsub_on_rx_refresh calls
subscription_persistence_update to apply the new expires timestamp.
This exposed other issues however...
* When creating a dialog from rdata (which sub_persistence_recreate
does from the packet buffer) there must NOT be a tag on the To
header (which there will be when a client refreshes a
subscription). If there is one, pjsip_dlg_create_uas will fail.
To address this, subscription_persistence_update now accepts a flag
that indicates that the original packet buffer must not be updated.
New subscribes don't set the flag and renews do. This makes sure
that when the rdata is recreated on asterisk startup, it's done
from the original subscribe packet which won't have the tag on To.
* When creating a dialog from rdata, we were setting the dialog's
remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
When the client tried to resubscribe after a restart with the
correct cseq, we'd reject the request with an Invalid CSeq error.
* The acts of creating a dialog and evsub by themselves when
recreating a subscription does NOT restart pjproject's subscription
timer. The result was that even if we did correctly recreate the
subscription, we never removed it if the client happened to go away
or send a non-OK response to a NOTIFY. However, there is no
pjproject function exposed to just set the timer on an evsub that
wasn't created by an incoming subscribe request. To address this,
we create our own timer using ast_sip_schedule_task. This timer is
used only for re-establishing subscriptions after a restart.
An earlier approach was to add support for setting pjproject's
timer (via a pjproject patch) and while that patch is still included
here, we don't use that call at the moment.
While addressing these issues, additional debugging was added and
some existing messages made more useful. A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.
ASTERISK-26696
ASTERISK-26756
Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e
This reverts commit 6492e91392.
The change in question was intended to prevent the need to reload in
order to update qualifies on contacts when an AOR changes. However, this
ended up causing a deadlock instead.
Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e
Prior to this change, qualifies would only update in the following
cases:
* A reload of res_pjsip.so was issued.
* A dynamic contact was re-registered after its AOR's qualify_frequency
had been changed
This does not work well if you are using realtime for your AORs. You can
update your database to have a new qualify_frequency, but the permanent
contacts on that AOR will not have their qualifies updated. And the
dynamic contacts on that AOR will not have their qualifies updated until
the next registration, which could be a long time.
This change seeks to fix this problem by making it so that whenever AOR
configuration is applied, the contacts pertaining to that AOR have their
qualifies updated.
Additions from this patch:
* AOR sorcery objects now have an apply handler that calls into a newly
added function in the OPTIONS code. This causes all contacts
associated with that AOR to re-schedule qualifies.
* When it is time to qualify a contact, the OPTIONS code checks to see
if the AOR can still be retrieved. If not, then qualification is
canceled on the contact.
Alterations from this patch:
* The registrar code no longer updates contact's qualify_frequence and
qualify_timeout. There is no point to this since those values already
get updated when the AOR changes.
* Reloading res_pjsip.so no longer calls the OPTIONS initialization
function. Reloading res_pjsip.so results in re-loading AORs, which
results in re-scheduling qualifies.
Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121
If endpoint ACLs were specified, they were not being freed
when endpoints were destroyed. On systems with realtime endpoints, this
could add up quickly since each DB lookup would allocate the ACL without
freeing it.
ASTERISK-26731 #close
Reported by Ustinov Artem
Change-Id: Ie1f8bf5b7a0de628c975beba01e69c56893331ad
Consider reviewing the expression of the 'A = B != C' kind.
The expression is calculated as following: 'A = (B != C)'
Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d
The PJSIPShowRegistrationsInbound AMI command was just dumping out
all AORs which was pretty useless and resource heavy since it had
to get all endpoints, then all aors for each endpoint, then all
contacts for each aor.
PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail
events which meets the intended purpose of the other command and has
significantly less overhead. Also, some additional fields that were
added to Contact since the original creation of the ContactStatusDetail
event have been added to the end of the event.
For compatibility purposes, PJSIPShowRegistrationsInbound is left
intact.
ASTERISK-26644 #close
Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a
Fix support of OS's like openBSD that use an older nameser.h,
this change reverts the defines to the older style which on other
systems is found in nameser_compat.h
Tested on openBSD 6.0, Debian 8
ASTERISK-26608 #close
Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a
This change fixes the SIP resolver such that if an IPv6 transport
is explicitly used it will resolve NAPTR, SRV, and AAAA records.
You can explicitly use one by specifying it on an endpoint.
ASTERISK-26571
Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.
ASTERISK-26309 #close
Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
The code was incorrectly invoking the unidentified logic when
an endpoint had actually been identified, causing log messages
to be output.
ASTERISK-26349 #close
Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f
This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
The endpoint identification PJSIP module is intended to identify which
endpoint an incoming request is from. If an endpoint is not identified,
then an artificial endpoint is used in its place when proceeding.
The problem is that the ACK request type is an exception to the rule.
The artificial endpoint is not used when processing an ACK. This results
in the possibility of having a NULL endpoint being used further on.
The reason ACK is an exception is an attempt not to spam security logs
with unidentified requests. Presumably, you've already logged the
unidentified request on the preceeding INVITE.
Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion
didn't cause an issue. A new change in 13.10 added endpoint ACL checking
shortly after endpoint identification. Because we are accessing a NULL
endpoint, this ACL check resulted in a crash.
The fix here is to be sure to retrieve the artificial endpoint for all
request types. ACKs still do not generate unidentified request security
events.
ASTERISK-26264 #close
Reported by nappsoft
AST-2016-006
Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703
Currently when you add global headers from the dialplan both
the header in the dialplan and the globally configured header
are added to the resulting SIP INVITE. This change makes it
so the headers in the dialplan take precedence and are the
only ones added.
Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
If the PJSIP endpoint's AOR with the permanent contact
was deleted from the realtime storage the res_pjsip module
continues trying to qualify this contact.
The error 'Unable to find an endpoint to qualify contact'
appeares every 'qualify_frequency' seconds.
This patch deletes this contact in this case.
The PJSIP endpoint's AOR with the permanent contact
is never qualified if it is added to realtime storage
after asterisk started.
This patch adds qualifying for the AOR's permanent contacts
on the first handling of this AOR.
ASTERISK-26319 #close
Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe
A recent change attempted to optimize startup by not updating contact
status. Instead, code responsible for qualifying contacts updates the
status as it becomes known. The code even accounts for contacts/AORs
that are not set to be qualified.
The problem, though, is when there are no contacts associated with an
endpoint. A common case is when an endpoint is set to register its
contacts but has not done so yet. In this case, prior to registration,
the endpoint's device state will appear to be "not in use" and hints
associated with that device will appear to be "idle". In actuality, the
device state and hint should both appear as "unavailable". The reason
for the failure is that the optimization change made all persistent
endpoint states set to "unknown".
The fix here is to change the hard-coded "unknown" to be "offline"
instead. The default state will be offline until the qualifying code
determines that the contact is actually online. This way, if there are
no contacts at all, then the state stays as offline, and device state
and hints appear correctly.
ASTERISK-26269 #close
Reported by nappsoft
Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a
We may check a global config option hundreds of times a second or more.
Asking sorcery for the global configuration from the config files backend
involves several allocations and container traversals. Using realtime
without a memory cache is a lot worse because you have to lookup in the
realtime database each time to reconstitute the sorcery object. With a
memory cache for realtime, there is about the same amount of overhead as
for config files. Either way, it is still fairly expensive to access the
sorcery object that much.
* Cache the global config options so we can access them faster. You must
now always perform a res_pjsip reload to change the global options.
Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7
The compilation failed for devmode
--enable DONT_OPTIMIZE
--enable BETTER_BACKTRACES
--enable DO_CRASH
--enable TEST_FRAMEWORK
res_pjsip/pjsip_configuration.c: In function dtls_handler:
res_pjsip/pjsip_configuration.c:974:20: error:
back may be used uninitialized in this function [-Werror=maybe-uninitialized]
int size = strlen(front);
^
cc1: all warnings being treated as errors
Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580
contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
If debug was specified in the global configuration but left blank,
the logger would treat it as a wildcard and log all hosts. If
default_from_user was empty, a crash would result.
The global apply handler now checks for empty strings.
ASTERISK-26239 #close
ASTERISK-26238 #close
Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336
* Eliminated RAII_VAR() usage in
ast_sip_persistent_endpoint_update_state().
* Added a missing allocation failure check to
persistent_endpoint_find_or_create().
* Made persistent_endpoint_find_or_create() create the new object without
a lock as it isn't needed.
* Cleaned up some ao2 container allocation idioms.
* Reordered res_pjsip_mwi.c load_module() and unload_module()
Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8
* Eliminated most RAII_VAR() usage.
* Added several missing allocation failure checks.
* Made ast_sip_for_each_contact() allocate the wrapper ao2 object without
a lock as it is not needed.
Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc
The named aor lock was always being locked for writes so a rwlock adds no
benefit and may be slower because rwlocks are biased toward read locking.
Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28
The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.
This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.
This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.
ASTERISK-26230 #close
Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a
The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
aor_observer_deleted() needs to operate on all contacts found for the
deleted AOR instead of only the first one found. This is really only a
problem if there is more than one contact for the AOR.
Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1
* Fix some whitespace in various routines.
* Rename i to iter in persistent_endpoint_update_state().
* Fix off-nominal copy/paste message wording in
persistent_endpoint_contact_deleted_observer()
Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8
The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.
1) It restarted any OPTIONS RTT ping cycle.
2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.
3) It cleared the RTT time each time the endpoint was refreshed.
4) The cleared RTT time was sent out as a statsd update each time.
5) It created two AMI events for each update.
* Revert the original patch and reimplement it. Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration. The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.
ASTERISK-26160 #close
Reported by: Matt Jordan
Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
When using TCP transport with chan_pjsip, the TCP_NODELAY
option value was allocated on the stack, then passed as a
pointer to the tcp transport configuration structure, and
later re-used on subsequently created sockets when it was
no longer valid. This patch changes the allocation to be
a static.
ASTERISK-26180 #close
Reported by: Scott Griepentrog
Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0
If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
ASTERISK-25471 #close
Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.
The status of endpoints with qualified aors will be updated by 'qualify'
functions.
ASTERISK-26061 #close
Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df
A non-existent constraint was being referenced in the upgrade script.
This patch corrects the problem by removing the reference.
In addition, the head of the alembic branch referred to a non-existent
revision. This has been fixed by referring to the proper revision.
This patch fixes another realtime problem as well. Our Alembic scripts
store booleans as yes or no values. However, Sorcery tries to insert
"true" or "false" instead. This patch introduces a new boolean type that
translates to "yes" or "no" instead.
ASTERISK-26128 #close
Change-Id: I51574736a881189de695a824883a18d66a52dcef
Sorcery creates taskprocessors for object types to process object observer
callbacks. An API call is needed to be able to set the congestion levels
of these taskprocessors for selected object types.
* Updated PJSIP's contact and contact_status sorcery object type observer
default congestion levels based upon stress testing. Increased the
congestion levels to reduce the potential for bursty register/unregister
and subscribe/unsubscribe activity from triggering the taskprocessor
overload alert.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6
When taskprocessors get backed up, there is a good chance that we are
being overloaded and need to defer adding new work to the system.
* Implemented a high/low water alert mechanism for modules to check if the
system is being overloaded and take appropriate action. When a
taskprocessor is created it has default congestion levels set. A
taskprocessor can later have those congestion levels altered for specific
needs if stress testing shows that the taskprocessor is a symptom of
overloading or needs to handle bursty activity without triggering an
overload alert.
* Add CLI "core show taskprocessor" low/high water columns.
* Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was
never a good thing to use when creating a taskprocessor because of the
nature of how its references needed to be cleaned up on a partial
creation.
* Made res_pjsip's distributor check if the taskprocessor overload alert
is active before placing a message representing brand new work onto a
distributor serializer.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I182f1be603529cd665958661c4c05ff9901825fa
Incoming messages that are not part of a dialog or a recognized response
to one of our requests need to be sent to a consistent serializer. Under
load we may be queueing retransmissions before we can process the original
message. We don't need to throw these messages onto random serializers
and cause reentrancy and message sequencing problems.
* Created a pool of pjsip/distributor serializers that get picked by
hashing the call-id and remote tag strings of the received messages.
* Made ast_sip_destroy_distributor() destroy items in the reverse order of
creation.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I2ce769389fc060d9f379977f559026fbcb632407
We should not be processing any incoming messages until we are fully
booted. We may not have dialplan or other needed configuration loaded
yet.
ASTERISK-26089 #close
Reported by: Scott Griepentrog
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264
The pjproject doxygen for rdata->msg_info.info says to call
pjsip_rx_data_get_info() instead of accessing the struct member directly.
You need to call the function mostly because the function will generate
the struct member value if it is not already setup.
Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799
As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.
Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.
ASTERISK-26011
Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
There are a lot of verbose messages about Endpoint and Contact status
changes if there are many dynamic endpoints.
The patch sets verbose level 2 for Endpoint status changes
and verbose level 3 for Contact status changes.
ASTERISK-26055 #close
Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7
The pjproject doxygen for rdata->msg_info.info says to call
pjsip_rx_data_get_info() instead of accessing the struct member directly.
You need to call the function mostly because the function will generate
the struct member value if it is not already setup.
Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2
When receiving an incoming response to a dialog-starting INVITE, we were
not matching the response to the INVITE dialog. Since we had not
recorded the to-tag to the dialog structure, the PJSIP-provided method
to find the dialog did not match.
Most of the time, this was not a problem, because there is a fall-back
that makes the response get routed to the same serializer that the
request was sent on. However, in cases where an asynchronous DNS lookup
occurs in the PJSIP core, the thread that sends the INVITE is not
actually a threadpool serializer thread. This means we are unable to
record a serializer to handle the incoming response.
Now, imagine what happens when an INVITE is sent on a non-serialized
thread, and an error response (such as a 486) arrives. The 486 ends up
getting put on some random threadpool thread. Eventually, a hangup task
gets queued on the INVITE dialog serializer. Since the 486 is being
handled on a different thread, the hangup task can execute at the same
time that the 486 is being handled. The hangup task assumes that it is
the sole owner of the INVITE session and channel, so it ends up
potentially freeing the channel and NULLing the session's channel
pointer. The thread handling the 486 can crash as a result.
This change has the incoming response match the INVITE transaction, and
then get the dialog from that transaction. It's the same method we had
been using for matching incoming CANCEL requests. By doing this, we get
the INVITE dialog and can ensure that the 486 response ends up being
handled by the same thread as the hangup, ensuring that the hangup runs
after the 486 has been completely handled.
ASTERISK-25941 #close
Reported by Javier Riveros
Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0
Although it's perfectly legal to place multiple SIP messages in the same packet,
it can cause problems because the Linux default is to enable Path MTU Discovery
which sets the Don't Fragment bit on the packets. If adding a second message to
the packet causes the MTU to be exceeded, and the destination isn't equipped to
send a FRAGMENTATION NEEDED response to a large packet, the packet will just be
dropped.
We can't specifically tell the stack to send only 1 message per packet, but we
can turn on TCP_NODELAY when we create the transport. This will at least tell
the stack to send packets as soon as possible.
ASTERISK-26005 #close
Reported-by: Ross Beer
Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd
With the old SIP module we can use IP access controls per peer.
PJSIP module missing this feature.
This patch added next configuration Endpoint options:
"acl" - list of IP ACL section names in acl.conf
"deny" - List of IP addresses to deny access from
"permit" - List of IP addresses to permit access from
"contact_acl" - List of Contact ACL section names in acl.conf
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
This patch also better logging failed request:
add custom message instead of "No matching endpoint found"
add SIP method to logging
ASTERISK-25900
Change-Id: I456dea3909d929d413864fb347d28578415ebf02
The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2.
SSL is not allowed. So, even if you specify "sslv3" for a transport method,
it's silently ignored and one of the TLS protocols is used. This was a new
behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that
we never caught.
Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default().
This tells pjproject to set the socket protocol to match the method.
ASTERISK-26004 #close
Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078
This patch modified pjsip_options to retrieve only
permament contacts for aor if the qualify_frequency is > 0
and persisted contacts if the qualify_frequency is > 0.
This patch also fixed a bug in res_sorcery_astdb.
res_sorcery_astdb doesn't save object data retrived from astdb.
ASTERISK-25826
Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05
With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.
With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing
This patch added contact.updated event.
ASTERISK-25904
Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate
multi-server setups.
ASTERISK-25931
Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8
A feature of chan_sip that service providers relied upon was the ability to
identify by the Authorization username. This is most often used when customers
have a PBX that needs to register rather than identify by IP address. From my
own experiance, this is pretty common with small businesses who otherwise
don't need a static IP.
In this scenario, a register from the customer's PBX may succeed because From
will usually contain the PBXs account id but an INVITE will contain the caller
id. With nothing recognizable in From, the service provider's Asterisk can
never match to an endpoint and the INVITE just stays unauthorized.
The fixes:
A new value "auth_username" has been added to endpoint/identify_by that
will use the username and digest fields in the Authorization header
instead of username and domain in the the From header to match an endpoint,
or the To header to match an aor. This code as added to
res_pjsip_endpoint_identifier_user rather than creating a new module.
Although identify_by was always a comma-separated list, there was only
1 choice so order wasn't preserved. So to keep the order, a vector was added
to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar
to find the aor. The res_pjsip_endpoint_identifier_* modules are called in
globals/endpoint_identifier_order.
Along the way, the logic in res_pjsip_registrar was corrected to match
most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
The order is:
username@domain
username@domain_alias
username
Auth by username does present 1 problem however, the first INVITE won't have
an Authorization header so the distributor, not finding a match on anything,
sends a securty_alert. It still sends a 401 with a challenge so the next
INVITE will have the Authorization header and presumably succeed. As a result
though, that first security alert is actually a false alarm.
To address this, a new feature has been added to pjsip_distributor that keeps
track of unidentified requests and only sends the security alert if a
configurable number of unidentified requests come from the same IP in a
configurable amout of time. Those configuration options have been added to
the global config object. This feature is only used when auth_username
is enabled.
Finally, default_realm was added to the globals object to replace the hard
coded "asterisk" used when an endpoint is not yet identified.
The testsuite tests all pass but new tests are forthcoming for this new
feature.
ASTERISK-25835 #close
Reported-by: Ross Beer
Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
This patch added new global pjsip option 'disable_multi_domain'.
Disabling Multi Domain can improve Realtime performance by reducing
number of database requests.
ASTERISK-25930 #close
Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7
The PJSIP parsing functions provide a nice concise way to check the
length of a hostname in a SIP URI. The problem is that in order to use
those parsing functions, it's required to use them from a thread that
has registered with PJLib.
On startup, when parsing AOR configuration, the permanent URI handler
may not be run from a PJLib-registered thread. Specifically, this could
happen when Asterisk was started in daemon mode rather than
console-mode. If PJProject were compiled with assertions enabled, then
this would cause Asterisk to crash on startup.
The solution presented here is to do our own parsing of the contact URI
in order to ensure that the hostname in the URI is not too long. The
parsing does not attempt to perform a full SIP URI parse/validation,
since the hostname in the URI is what is important.
ASTERISK-25928 #close
Reported by Joshua Colp
Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
There are several places that do scheduled tasks or periodic housecleaning,
each with its own implementation:
* res_pjsip_keepalive has a thread that sends keepalives.
* pjsip_distributor has a thread that cleans up expired unidentified requests.
* res_pjsip_registrar_expire has a thread that cleans up expired contacts.
* res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task.
* res_pjsip_sdp_rtp also uses ast_sched to send keepalives.
There are also places where we should be doing scheduled work but aren't.
A good example are the places we have sorcery observers to start registration
or qualify. These don't work when changes are made to a backend database
without a pjsip reload. We need to check periodically.
As a first step to solving these issues, a new ast_sip_sched facility has
been created.
ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue.
When a task is ready to run, ast_sip_task_pusk is called for it. This ensures
that the task is executed in a PJLIB registered thread and doesn't hold up the
ast_sched thread so it can immediately continue processing the queue. The
serializer used by ast_sip_sched is one of your choosing or a random one from
the res_pjsip pool if you don't choose one.
Another feature is the ability to automatically clean up the task_data when the
task expires (if ever). If it's an ao2 object, it will be dereferenced, if
it's a malloc'd object it will be freed. This is selectable when the task is
scheduled. Even if you choose to not auto dereference an ao2 task data object,
the scheduler itself maintains a reference to it while the task is under it's
control. This prevents the data from disappearing out from under the task.
There are two scheduling models.
AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at
the specific interval. That is, every "interval" milliseconds, regardless of
how long the task takes. If the task takes longer than the interval, it will
be scheduled at the next available multiple of interval. For exmaple: If the
task has an interval of 60 secs and the task takes 70 secs (it better not),
the next invocation will happen at 120 seconds.
AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should
start "interval" milliseconds after the current invocation has finished.
Also, the same ast_sched facility for fixed or variable intervals exists. The
task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or
AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time.
One res_pjsip.h housekeeping change was made. The pjsip header files were
added to the top. There have been a few cases lately where I've needed
res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because
I didn't add the pjsip header files to my source even though I never referenced
any pjsip calls.
Finally, a few new convenience APIs were added to astobj2 to make things a
little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and
ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros
were also copied from res_phoneprov because I got tired of having to duplicate
the same hash, sort and compare functions over and over again. The
AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for
aor_container_alloc into your source.
This facility can be used immediately for the situations where we already have
a thread that wakes up periodically or do some scheduled work. For the
registration and qualify issues, additional sorcery and schema changes would
need to be made so that we can easily detect changed objects on a periodic
basis without having to pull the entire database back to check. I'm thinking
of a last-updated timestamp on the rows but more on this later.
Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c
Due to some ignored return values, Asterisk could crash if processing an
incoming REGISTER whose contact URI was above a certain length.
ASTERISK-25707 #close
Reported by George Joseph
Patches:
0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
AST-2016-004
Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d
* Added Useragent and RegExpire headers to AMI Event
ContactStatusDetail with associated documentation.
ASTERISK-25903 #close
Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239
Contact expiration can occur in several places: res_pjsip_registrar,
res_pjsip_registrar_expire, and automatically when anyone calls
ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar
may also be attempting to renew or add a contact. Since none of this was locked
it was possible for one thread to be renewing a contact and another thread to
expire it immediately because it was working off of stale data. This was the
casue of intermittent registration/inbound/nominal/multiple_contacts test
failures.
Now, the new named lock functionality is used to lock the aor during contact
expire and add operations and res_pjsip_registrar_expire now checks the
expiration with the lock held before deleting the contact.
ASTERISK-25885 #close
Reported-by: Josh Colp
Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059
Mark Michelson
Alexei Gradinari
Joshua Colp
George Joseph
Richard Mudgett
Richard Begg
Jørgen H
zuul
Badalyan Vyacheslav
snuffy
Corey Farrell
Aaron An
Scott Griepentrog
George Joseph