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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
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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
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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
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contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
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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
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Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes
on endpoints for unsolicited mwi and on aors for subscriptions required
that the admin know in advance which the client wanted. If you specified
mailboxes on the endpoint, subscriptions were rejected even if you also
specified mailboxes on the aor.
Voicemail extension:
* Added a global default_voicemail_extension which defaults to "".
* Added voicemail_extension to both endpoint and aor.
* Added ast_sip_subscription_get_dialog for support.
* Added ast_sip_subscription_get_sip_uri for support.
When an unsolicited NOTIFY is constructed, the From header is parsed, the
voicemail extension from the endpoint is substituted for the user, and the
result placed in the Message-Account field in the body.
When a subscribed NOTIFY is constructed, the subscription dialog local uri
is parsed, the voicemail_extension from the aor (looked up from the
subscription resource name) is substituted for the user, and the result
placed in the Message-Account field in the body.
If no voicemail extension was defined, the Message-Account field is not added
to the NOTIFY body.
mwi_subscribe_replaces_unsolicited:
* Added mwi_subscribe_replaces_unsolicited to endpoint.
The previous behavior was to reject a subscribe if a previous internal
subscription for unsolicited MWI was found for the mailbox. That remains the
default. However, if there are mailboxes also set on the aor and the client
subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
subscription is removed and replaced with the external subscription. This
allows an admin to configure mailboxes on both the endpoint and aor and allows
the client to select which to use.
ASTERISK-25865 #close
Reported-by: Ross Beer
Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea
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There were a number of places in the res_pjsip stack that were getting
all endpoints or all aors, and then filtering them locally.
A good example is pjsip_options which, on startup, retrieves all
endpoints, then the aors for those endpoints, then tests the aors to see
if the qualify_frequency is > 0. One issue was that it never did
anything with the endpoints other than retrieve the aors so we probably
could have skipped a step and just retrieved all aors. But nevermind.
This worked reasonably well with local config files but with a realtime
backend and thousands of objects, this was a nightmare. The issue
really boiled down to the fact that while realtime supports predicates
that are passed to the database engine, the non-realtime sorcery
backends didn't.
They do now.
The realtime engines have a scheme for doing simple comparisons. They
take in an ast_variable (or list) for matching, and the name of each
variable can contain an operator. For instance, a name of
"qualify_frequency >" and a value of "0" would create a SQL predicate
that looks like "where qualify_frequency > '0'". If there's no operator
after the name, the engines add an '=' so a simple name of
"qualify_frequency" and a value of "10" would return exact matches.
The non-realtime backends decide whether to include an object in a
result set by calling ast_sorcery_changeset_create on every object in
the internal container. However, ast_sorcery_changeset_create only does
exact string matches though so a name of "qualify_frequency >" and a
value of "0" returns nothing because the literal "qualify_frequency >"
doesn't match any name in the objset set.
So, the real task was to create a generic string matcher that can take a
left value, operator and a right value and perform the match. To that
end, strings.c has a new ast_strings_match(left, operator, right)
function. Left and right are the strings to operate on and the operator
can be a string containing any of the following: = (or NULL or ""), !=,
>, >=, <, <=, like or regex. If the operator is like or regex, the
right string should be a %-pattern or a regex expression. If both left
and right can be converted to float, then a numeric comparison is
performed, otherwise a string comparison is performed.
To use this new function on ast_variables, 2 new functions were added to
config.c. One that compares 2 ast_variables, and one that compares 2
ast_variable lists. The former is useful when you want to compare 2
ast_variables that happen to be in a list but don't want to traverse the
list. The latter will traverse the right list and return true if all
the variables in it match the left list.
Now, the backends' fields_cmp functions call ast_variable_lists_match
instead of ast_sorcery_changeset_create and they can now process the
same syntax as the realtime engines. The realtime backend just passes
the variable list unaltered to the engine. The only gotcha is that
there's no common realtime engine support for regex so that's been noted
in the api docs for ast_sorcery_retrieve_by_fields.
Only one more change to sorcery was done... A new config flag
"allow_unqualified_fetch" was added to reg_sorcery_realtime.
"no": ignore fetches if no predicate fields were supplied.
"error": same as no but emit an error. (good for testing)
"yes": allow (the default);
"warn": allow but emit a warning. (good for testing)
Now on to res_pjsip...
pjsip_options was modified to retrieve aors with qualify_frequency > 0
rather than all endpoints then all aors. Not only was this a big
improvement in realtime retrieval but even for config files there's an
improvement because we're not going through endpoints anymore.
res_pjsip_mwi was modified to retieve only endpoints with something in
the mailboxes field instead of all endpoints then testing mailboxes.
res_pjsip_registrar_expire was completely refactored. It was retrieving
all contacts then setting up scheduler entries to check for expiration.
Now, it's a single thread (like keepalive) that periodically retrieves
only contacts whose expiration time is < now and deletes them. A new
contact_expiration_check_interval was added to global with a default of
30 seconds.
Ross Beer reports that with this patch, his Asterisk startup time dropped
from around an hour to under 30 seconds.
There are still objects that can't be filtered at the database like
identifies, transports, and registrations. These are not going to be
anywhere near as numerous as endpoints, aors, auths, contacts however.
Back to allow_unqualified_fetch. If this is set to yes and you have a
very large number of objects in the database, the pjsip CLI commands
will attempt to retrive ALL of them if not qualified with a LIKE.
Worse, if you type "pjsip show endpoint <tab>" guess what's going to
happen? :) Having a cache helps but all the objects will have to be
retrieved at least once to fill the cache. Setting
allow_unqualified_fetch=no prevents the mass retrieve and should be used
on endpoints, auths, aors, and contacts. It should NOT be used for
identifies, registrations and transports since these MUST be
retrieved in bulk.
Example sorcery.conf:
[res_pjsip]
endpoint=config,pjsip.conf,criteria=type=endpoint
endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error
ASTERISK-25826 #close
Reported-by: Ross Beer
Tested-by: Ross Beer
Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67
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Per RFC3325, the 'From' header is now anonymized on outgoing calls when
caller id presentation is prohibited.
TID = trust_id_outbound
PRO = Set(CALLERID(pres)=prohib)
USR = endpoint/from_user
DOM = endpoint/from_domain
PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes)
Conditions |Result
--------------------|----------------------------------------------------
TID PRO USR DOM |PAI FROM
--------------------|----------------------------------------------------
Y Y abc def.ghi |PRI "Anonymous" <sip:abc@def.ghi>
Y Y abc |PRI "Anonymous" <sip:abc@anonymous.invalid>
Y Y def.ghi |PRI "Anonymous" <sip:anonymous@def.ghi>
Y Y |PRI "Anonymous" <sip:anonymous@anonymous.invalid>
Y N abc def.ghi |YES <sip:abc@def.ghi>
Y N abc |YES <sip:abc@<ip_address>>
Y N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi>
Y N |YES "Caller Name" <sip:<caller_exten>@<ip_address>>
N Y abc def.ghi |NO "Anonymous" <sip:abc@def.ghi>
N Y abc |NO "Anonymous" <sip:abc@anonymous.invalid>
N Y def.ghi |NO "Anonymous" <sip:anonymous@def.ghi>
N Y |NO "Anonymous" <sip:anonymous@anonymous.invalid>
N N abc def.ghi |YES <sip:abc@def.ghi>
N N abc |YES <sip:abc@<ip_address>>
N N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi>
N N |YES "Caller Name" <sip:<caller_exten>@<ip_address>>
ASTERISK-25791 #close
Reported-by: Anthony Messina
Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9
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The 'reload' mechanism actually involves closing the underlying
socket and calling the appropriate udp, tcp or tls start functions
again. Only outbound_registration, pubsub and session needed work
to reset the transport before sending requests to insure that the
pjsip transport didn't get pulled out from under them.
In my testing, no calls were dropped when a transport was changed
for any of the 3 transport types even if ip addresses or ports were
changed. To be on the safe side however, a new transport option was
added (allow_reload) which defaults to 'no'. Unless it's explicitly
set to 'yes' for a transport, changes to that transport will be ignored
on a reload of res_pjsip. This should preserve the current behavior.
Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf
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Attempting to load a transport from realtime was forcing asterisk into an
infinite recursion loop. The first thing transport_apply did was to do a
sorcery retrieve by id for an existing transport of the same name. For files,
this just returns the previous object from res_sorcery_config's internal
container, if any. For realtime, the res_sourcery_realtime driver looks in the
database and finds the existing row but now it has to rehydrate it into a
sorcery object which means calling... transport_apply. And so it goes.
The main issue with loading from realtime (apart from the loop) was that
transport stores structures and pointers directly in the ast_sip_transport
structure instead of the separate ast_transport_state structure. This patch
separates those items into the ast_sip_transport_state structure. The pattern
is roughly the same as res_pjsip_outbound_registration.
Although all current usages of ast_sip_transport and ast_sip_transport_state
were modified to use the new ast_sip_get_transport_state API, the original
items are left in ast_sip_transport and kept updated to maintain ABI
compatability for third-party modules. They are marked as deprecated and
noted that they're now in ast_sip_transport_state.
ASTERISK-25606 #close
Reported-by: Martin Moučka
Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19
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Added new global option (regcontext) to pjsip. When set, Asterisk will
dynamically create and destroy a NoOp priority 1 extension
for a given endpoint who registers or unregisters with us.
ASTERISK-25670 #close
Reported-by: Daniel Journo
Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62
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On a system with multiple ip addresses in the same subnet, if a
transport is bound to a specific ip address and endpoint/media_address
is set, the SIP/SDP will have the correct address in all fields but
the rtp stream MAY still originate from one of the other ip addresses,
most probably the "primary" ip address. This happens because
res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
the "all" ip address (0.0.0.0 or ::).
The new option causes res_pjsip_sdp_rtp/create_rtp to call
ast_rtp_instance_new with the endpoint's media_address (if specified)
instead of the "all" address. This causes the packets to originate from
the specified address.
ASTERISK-25632
ASTERISK-25637
Reported-by: Olivier Krief
Reported-by: Dan Journo
Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
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PJSIP name formats:
pjsip/aor/<aor>-<seq> -- registrar thread pool serializer
pjsip/default-<seq> -- default thread pool serializer
pjsip/messaging -- messaging thread pool serializer
pjsip/outreg/<registration>-<seq> -- outbound registration thread pool
serializer
pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer
pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer
pjsip/session/<endpoint>-<seq> -- session thread pool serializer
pjsip/websocket-<seq> -- websocket thread pool serializer
Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084
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An earlier commit changed the id of dynamic contacts to contain
a hash instead of the uri. This patch updates status change
logging to show the aor/uri instead of the id. This required
adding the aor id to contact and contact_status and adding
uri to contact_status. The aor id gets added to contact and
contact_status in their allocators and the uri gets added to
contact_status in pjsip_options when the contact_status is
created or updated.
ASTERISK-25598 #close
Reported-by: George Joseph
Tested-by: George Joseph
Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511
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We have observed situations where the SIP threadpool may become
deadlocked. However, because incoming traffic is still arriving, the SIP
threadpool's queue can continue to grow, eventually running the system
out of memory.
This change makes it so that incoming traffic gets rejected with a 503
response if the queue is backed up too much.
Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816
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In a realtime based system with a limited number of threadpool threads
it is possible for a deadlock to occur. This happens when permanent
endpoint state is updated, which will cause database queries to be done.
These queries may result in URI validation being done which is done
synchronously using a PJSIP thread. If all PJSIP threads are in use
processing traffic they themselves may be blocked waiting to get the
permanent endpoint container lock when identifying an endpoint.
This change moves URI validation to occur at use time instead of
configuration time. While this comes at a cost of not seeing a problem
until you use it it does solve the underlying deadlock problem.
ASTERISK-25486 #close
Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a
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The default_from_user retrieval function was pulling the
default_from_user from the global configuration struct in an unsafe way.
If using a database as a backend configuration store, the global
configuration struct is short-lived, so grabbing a pointer from it
results in referencing freed memory.
The fix here is to copy the default_from_user value out of the global
configuration struct.
Thanks go to John Hardin for discovering this problem and proposing the
patch on which this fix is based.
ASTERISK-25390 #close
Reported by Mark Michelson
Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c
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When Asterisk sends an outbound SIP request, if there is no direct
reason to place a specific value for the username in the From header,
Asterisk would generate a UUID. For example, this would happen when
sending outbound OPTIONS requests when qualifying or when sending
outbound INVITE requests when originating (if no explicit caller ID were
provided). The issue is that some SIP providers reject these sorts of
requests with a "Name too long" error response.
This patch aims to fix this by changing the default outbound username in
From headers to "asterisk". This value can be overridden by changing the
default_from_user option in the global options if desired.
ASTERISK-25377 #close
Reported by Mark Michelson
Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190
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Modules commonly used the pj_gethostip function for retrieving the
IP address of the host. This function does not cache the result and may
result in a DNS lookup occurring, or additional work. If the DNS
server is unreachable or network issues arise this can cause the
pj_gethostip function to block for a period of time.
This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
function which does the same thing but caches the host IP address at
module load time. This results in no additional work being done each
time the local host IP address is needed.
ASTERISK-25342 #close
Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e
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This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
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This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.
ASTERISK-25242 #close
Reported by Mark Michelson
Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
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Previously Asterisk did not properly failover to the next resolved DNS
address when a endpoint could not be reached. With this patch, and while
using res_pjsip, SIP requests (both in/out of dialog) now attempt to use
the next address in the list of resolved addresses until a proper response
is received or no more addresses are left.
ASTERISK-25076 #close
Reported by: Joshua Colp
Change-Id: Ief14f4ebd82474881f72f4538f4577f30af2a764
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A module trying to unload needs to wait for all serializers it creates and
uses to complete processing before unloading.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059
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The res_pjsip_mwi previously required a reload to set up the proper
subscriptions to allow unsolicited MWI to work. This change
makes it so the act of registering will also cause this to occur.
This is particularly useful if realtime is involved as no reload
needs to occur within Asterisk to cause the MWI information
to get sent.
ASTERISK-25180 #close
Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252
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Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
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Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976
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Add a new ContactStatus AMI event.
Publish the following status/state changes:
Created
Removed
Reachable
Unreachable
Unknown
Contact URI, new status/state, aor and endpoint names, and the
last qualify rtt result are included in the event.
ASTERISK-25114 #close
Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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This patch refactors the transaction timeout processing to eliminate
calling the lower level public pjsip functions and reverts to calling
pjsip_endpt_send_request again. This is the result of me noticing
a possible incompatibility with pjproject-2.4 which was causing
contact status flapping.
The original version of this feature used the lower level calls to
get access to the tsx structure in order to cancel the transaction
when our own timer expires. Since we no longer have that access,
if our own timer expires before the pjsip timer, we call the callbacks
and just let the pjsip transaction take it's own course. When the
transaction ends, it discovers the callbacks have already been run
and just cleans itself up.
A few messages in pjsip_configuration were also added/cleaned up.
ASTERISK-25105 #close
Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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Odd behaviors have been observed during outbound registrations. The most
common problem witnessed has been one where a request with
authentication credentials cannot be created after receiving a 401
response. Other behaviors include apparently processing an incorrect SIP
response.
Inspecting the code led to an apparent issue with regards to how we
handle transactions in outbound registration code. When a response to a
REGISTER arrives, we save a pointer to the transaction and then push a
task onto the registration serializer. Between the time that we save the
pointer and push the task, it's possible for the transaction to be
destroyed due to a timeout. It's also possible for the address to be
reused by the transaction layer for a new transaction.
To allow for authentication of a REGISTER request to be authenticated
after the transaction has timed out, we now hold a reference to the
original REGISTER request instead of the transaction. The function for
creating a request with authentication has been altered to take the
original request instead of the transaction where the original request
was sent.
ASTERISK-25020
Reported by Mark Michelson
Change-Id: I756c19ab05ada5d0503175db9676acf87c686d0a
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The "Add qualify_timeout processing and eventing" patch introduced
an issue where contacts that had qualify_frequency set to 0 were
showing Unavailable instead Unknown. This patch checks for
qualify_frequency=0 and create an "Unknown" contact_status
with an RTT = 0.
Previously, the lack of contact_status implied Unknown but since
we're now changing endpoint state based on contact_status, I've
had to add new UNKNOWN status so that changes could trigger the
appropriate contact_status observers.
ASTERISK-24977: #close
Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7
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Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup. So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.
This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies. This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.
If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random(). If not set,
qualify_timeout is used.
The default is "0" (disabled).
ASTERISK-24863 #close
Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>
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This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
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This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
Since we currently have no control over pjproject transaction timeout, this
patch pulls the pjsip_endpt_send_request function out of pjproject and into
res_pjsip/endpt_send_transaction in order to implement that capability.
Now when the transaction is initiated, we also schedule our own pj_timer with
our own desired timeout.
If the transaction completes before either timeout, pjproject cancels its timer,
and calls our tsx callback where we cancel our timer and run the app callback.
If the pjproject timer times out first, pjproject calls our tsx callback where
we cancel our timer and run the app callback.
If our timer times out first, we terminate the transaction which causes
pjproject to cancel its timer and call our tsx callback where we run the app
callback.
Regardless of the scenario, pjproject is calling the tsx callback inside the
group_lock and there are checks in the callback to make sure it doesn't run
twice.
As part of this patch ast_sip_send_out_of_dialog_request was created to replace
its similarly named private function. It takes a new timeout argument in
milliseconds (<= 0 to disable the timeout).
ASTERISK-24863 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747
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