HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered.
This patch change the call order of ast_queue_control_data
and ast_queue_control in chan_pjsip_incoming_response.
ASTERISK-26396 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Ide2d31723d8d425961e985de7de625694580be61
For the channel driver chan_sip, you specify externhost=example.com in sip.conf
when your Asterisk is behind a NAT and your IP address is assigned dynamically.
Or stated differently: You do not have a static IP address to use "externaddr"
directly. This NAT support is quite handy but just about IPv4. Previously,
Asterisk resolved "externhost" to any IP version. When the first DNS answer
resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and
connection (c=). This happened in outgoing SIP-REGISTER and while answering
SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an
IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost".
ASTERISK-18232 #close
Reported by: Jacek Kowalski
Tested by: Alexander Traud
patches:
changes.patch submitted by Alessandro Crespi
Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac
Users upgrading from asterisk 13.5 to a later version and who use
realtime with peers that have mailboxes were experiencing runaway
situations that manifested as a continuous stream of taskprocessor
congestion errors, memory leaks and an unresponsive chan_sip.
A related issue was that setting rtcachefriends=no NEVER worked in
asterisk 13 (since the move to stasis). In 13.5 and earlier, when a
peer tried to register, all of the stasis threads would block and
chan_sip would again become unresponsive. After 13.5, the runaway
would happen.
There were a number of causes...
* mwi_event_cb was (indirectly) calling build_peer even though calls to
mwi_event_cb are often caused by build_peer.
* In an effort to prevent chan_sip from being unloaded while messages
were still in flight, destroy_mailboxes was calling
stasis_unsubscribe_and_join but in some cases waited forever for the
final message.
* add_peer_mailboxes wasn't properly marking the existing mailboxes
on a peer as "keep" so build_peer would always delete them all.
* add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
then just creating them again.
All of this was causing a flood of subscribes and unsubscribes on
multiple threads all for the same peer and mailbox.
Fixes...
* add_peer_mailboxes now marks mailboxes correctly and build_peer only
deletes the ones that really are no longer needed by the peer.
* add_peer_mwi_subs now only adds subscriptions marked as "new" instead
of unsubscribing and resubscribing everything. It also adds the peer
object's address to the mailbox instead of its name to the subscription
userdata so mwi_event_cb doesn't have to call build_peer.
With these changes, with rtcachefriends=yes (the most common setting),
there are no leaks, locks, loops or crashes at shutdown.
rtcachefriends=no still causes leaks but at least it doesn't lock, loop
or crash. Since making rtcachefriends=no work wasnt in scope for this
issue, further work will have to be deferred to a separate patch.
Side fixes...
* The ast_lock_track structure had a member named "thread" which gdb
doesn't like since it conflicts with it's "thread" command. That
member was renamed to "thread_id".
ASTERISK-25468 #close
Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0
Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for
SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP
connections, allowing the TCP layer to handle the retransmits. Unfortunately,
this caused sessions to be terminated with a retransmit timeout becasue it
stopped at the point of the first retrans call.
This patch waits for the 64*T1 timer to expire instead.
ASTERISK-19968
Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204
(cherry picked from commit 98e42cc662)
Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
Session-Timers for SIP over TCP (and for SIP over TLS).
However with longer international calls via TCP, the SIP channel might break,
because all hops on the Internet route must stay online (have not a single power
outage, for example). Therefore with Session-Timers enabled (which are enabled
at default), you might see dropped calls. Consequently even with this change,
you might be better-off going for session-timers=refuse in your sip.conf.
ASTERISK-19968 #close
Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957
(cherry picked from commit 66c9dfb272)
Previously, the Contact was stored only on initial INVITE and on any
18X and 200. That meant that after re-INVITEs from *us* the Contact
could get updated, but after re-INVITEs from the *peer*, it did not.
This changeset fixes this inconsistency, properly allowing target
refreshes through re-INVITES (RFC3261, 12.2).
If your strictrtp setting allows it, this change allows you to switch
the source IP of a connected/calling device mid-call with a simple
re-INVITE from the new IP.
ASTERISK-26358 #close
Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435
In some scenarios dialog_initialize_rtp can be called multiple times on
the same dialog. This can cause RTP instances to be leaked along with
multiple file descriptors for each instance.
This change makes it so the existing RTP instances are destroyed and
not overwritten, stopping the memory leak.
ASTERISK-26272 #close
patches:
ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909)
Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73
Certain SNOM phones send so-called "optional crypto" in their SDP body.
Regular SRTP setup looks like this:
m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...
SNOM-style "optional crypto" looks like this:
m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...
A crypto line is supplied, but the m-line does not have SAVP.
When res_srtp.so is *not* loaded, then chan_sip.so treats the optional
crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the
incoming call with the following message:
WARNING: process_sdp: Failed to receive SDP offer/answer with
required SRTP crypto attributes for audio
For platforms that want to start providing SRTP this presents a
compatibility problem.
This changeset lets chan_sip handle the SDP as if no crypto-line was
supplied: i.e. accept the call as regular RTP, just like it did before
res_srtp was loaded.
Now you'll get this informative warning instead:
WARNING: Ignoring crypto attribute in SDP because RTP transport is
insecure
ASTERISK-23989 #close
Reported by: Olle Johansson
Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2
Historically, Asterisk has always specified annexb=no for the g729 format.
However, when using res_pjsip no format attribute was specified. This patch
makes it so the SDP now contains a format attribute line with annexb=no.
Note, that this means only g729a is negotiated. Even for pass through support.
According to rfc7261 the type of annex used (a or b) is dependent upon the
answerer. However, Asterisk being a back to back user agent makes this tricky
to support at this time, thus we only allow annex 'a' for now.
ASTERISK-26228 #close
patches:
res_format_attr_g729.c submitted by Jason Parker (license 4993)
Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0
This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver. Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel. Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.
Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea
Packets are read regulary, when there is no data in buffer fr->frametype
is AST_FRAME_NULL. There was no check of frametype and lastrtprx always
updated and, therefore, rtptimeout did not work at all.
ASTERISK-25270 #close
Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d
* Following the example of the PJSIP channel driver, the channel
technology specific documentation has been moved to the respective
channel drivers that provide that functionality. This has the benefit
of locating the documentation of items with those modules that provide
it.
* Examples of using the CHANNEL function for both standard items as well
as for PJSIP have been added.
* The 'max_forwards' standard item has been documented.
Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b
This patch adds a new PJSIP specific dialplan function,
PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
session will be refreshed via either an UPDATE or re-INVITE request.
When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
the formats in use on a PJSIP channel can be re-negotiated and changed
dynamically after call setup.
ASTERISK-26277 #close
Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b
The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to
deadlock if an incoming fax happens during the Playback or similar
application.
* Fixed the potential deadlock by not calling ast_async_goto() with the
channel lock held.
ASTERISK-26216 #close
Reported by: Richard Mudgett
Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa
The sip_read() has the potential to deadlock if an incoming fax happens
during the Playback or similar application.
* Fixed the potential deadlock by not calling ast_async_goto() with the
channel lock held.
* Made always eat the fax detection frame whether there is a fax extension
or not.
ASTERISK-26216
Reported by: Richard Mudgett
Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e
The chan_pjsip_cng_tone_detected() has the potential to deadlock if an
incoming fax happens during the Playback or similar application.
* Fixed the potential deadlock by not calling ast_async_goto() with the
channel lock held.
* Made always eat the fax detection frame whether there is a fax extension
or not.
ASTERISK-26216
Reported by: Richard Mudgett
Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5
The new option allows the channel driver's faxdetect 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.
* Don't clear dsp_features after passing them to the dsp code in
my_pri_ss7_open_media(). We should still remember them especially for the
new faxdetect_timeout option.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Ieffd3fe788788d56282844774365546dce8ac810
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
sip_show_channels locks the dialogs container first then locks each
sip_pvt so it can spit out the details. The rest of sip dialog
processing locks the sip_pvt first then locks the dialogs container
if it needs to. Both lock in the order they need but deadlocks can
result. To fix, sip_show_channels and sip_show_channelstats have
been converted to use an iterator rather than ao2_callback. This way
the container is locked only while getting the next entry and is
unlocked when the callback is called.
ASTERISK-23013 #close
Change-Id: Id9980419909e811f89484950ed46ef117b9eb990
If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead
code. This places all existing unit tests into a conditional block if
they weren't already.
ASTERISK-26211 #close
Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686
Cleanup the peer reference when stasis_subscription_final_message is
true. Also free peer_name even if peer exists, after reload a new
peer_name will be allocated.
ASTERISK-26193 #close
Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69
* get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error.
* build_peer leaks peer on failure to allocate the endpoint.
This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed
with an unref in the appropriate place.
ASTERISK-26184 #close
Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12
Some T.38 implementations may send another re-invite after the initial
one which adds additional negotiation details (such as the max bitrate).
Currently this will fail when passthrough is being done in chan_sip as we
do nothing if T.38 is already active.
Other handlers of T.38 inside of Asterisk (such as res_fax) handle this
scenario so this change adds support for it to chan_sip and res_pjsip_t38.
If a request to negotiate is received while T.38 is already enabled a
new re-INVITE is sent and negotiation is done again.
ASTERISK-26179 #close
Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c
pjsip_inv_end_session() is documented as being able to return the
passed in tdata parameter set to NULL on success.
Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047
gcc 6.1.1 caught a few more issues.
Made sure the unit tests still pass for the func_env and stdtime
issues.
ASTERISK-26157 #close
Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e
This change removes hardcoded SDP parsing and generation for
Siren7 and Siren14 from chan_sip and moves it to format attribute
modules so it can also be used by chan_pjsip.
With this the fmtp lines for both are added with the bitrate
information.
ASTERISK-26021
Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037
A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD)
was using a pointer to a pointer as the destination of a memcpy and a
'&' instead of '*' in the sizeof.
ASTERISK-26138 #close
Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708
bridge_native_rtp can call into an RTP-capable channel driver in order
for the driver to update information about who the channel is
communicating with. For SIP channel drivers, this means deactivating
RTCP and sending a reinvite so that the endpoints can communicate
directly.
bridge_native_rtp does the right thing and has the channel locked when
calling into the channel driver. chan_pjsip can't alter session
properties in this thread, though. chan_pjsip queues a task on the
session serializer in order to update properties there.
The problem is that this queued task was not locking the channel. This
meant that the queued task could attempt to deactivate RTCP at the same
time that the channel thread was attempting to process an incoming RTCP
packet. This could lead to a crash.
This patch fixes the issue by locking the channel in the queued task
when altering RTP properties.
ASTERISK-26092 #close
Reported by Niklas Larsson
Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159
A change to glibc 2.22 changed the order of the sockadddr_storage
members which caused the places where we do an initialization of
ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those
initializers (which we shouldn't have been using anyway) have been
replaced with memsets.
Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4
POSIX defines signal.h. sys/signal.h should not be used as it is
c-library internal header which may or may not exist. Notably with
musl it generates warning of being incorrect.
Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc
This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compactheaders=yes via the file sip.conf.
ASTERISK-25578 #close
Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044
Currently chan_sip can give weird messages if the contacts don't
fit in the From: or To: headers. This fix changes the from,to and
invite variables to use ast_str, allocates and deallocates them and
resizes them if needed.
ASTERISK-26069 #close
Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3
Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])
Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.
Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9
This adds a new parameter to the end of a multicast RTP dialing string.
This parameter defines the following options:
* i: Set the interface from which multicast RTP is sent
* l: Set whether multicast packets are looped back to the sender
* t: Set the TTL for multicast packets
* c: Set the codec to use for RTP
ASTERISK-26068 #close
Reported by Mark Michelson
Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
* Fixed NULL crash potential if parameters are missing.
* Reordered some operations so further diagnostic messages can be
more helpful.
Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70
When chan_sip does a re-INVITE to refresh a session and authentication
is required, the INVITE with the Authorization header containes a
second Session-Expires header without the ";refersher=" parameter.
This is causing some proxies to return a 400. Also, when Asterisk is
the uas and the refresher, it is including the Session-Expires and
Min-SE headers in OPTIONS messages which is not allowed per RFC4028.
This patch (based on the reporter's) Checks to see if a Session-Expires
header is already in the message before adding another one. It also
checks that the method is INVITE or UPDATE.
ASTERISK-26030 #close
Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9
The unload process currently tells each TCP/TLS to terminate but
does not wait for them to do so. This introduces a race condition
where the container holding the threads may be destroyed before
the threads are able to remove themselves from it. When they
finally do the container is invalid and can't be used causing a
crash.
A previous change existed which waited a bit to wait for any
stranglers to finish. This change extends this and waits longer.
ASTERISK-25961 #close
Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6
Since Stasis has been introduced, an attempt to send AMI messages by an
autocreated peer caused a crash, and all events from autocreated peers were
semi-inadvertently disabled altogether in 0b83761. This change restores the
disabled functionality.
ASTERISK-25950
Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974
This eliminates some casts that I made a note saying v10 and above
would no longer need them.
Better late than never :)
Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572
Added the ability to show channel statistics to chan_pjsip (cli_functions.c)
Moved the existing 'pjsip show channel(s)' functionality from
pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's
private header so it made sense to move the existing channel commands as well.
Now using stasis_cache_dump to get the channel snapshots rather than retrieving
all endpoints, then getting each one's channel snapshots. Much more efficient.
Change-Id: I03b114522126d27434030b285bf6d531ddd79869
Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those
codecs, which the caller did not request/support. That fix was not complete
because on the second Session Timer all codecs were sent again. Some VoIP/SIP
clients interpreted that complete codec-list as a change in the SIP session.
Because of that, Asterisk did not send the RTP audio via NAT anymore which
created a non-audio scenario after the second Session Timer fired.
ASTERISK-24543 #close
Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66
check_via() already skips leading blanks where the sent-by address (with the
optional port) should be placed.
Since RFC 3261 allows for blanks between the port ant the Via parameters:
> https://tools.ietf.org/html/rfc3261#section-20.42
(actually it allows a lot of blanks more ;-)). I just switched from
ast_skip_blanks() to ast_strip() on the local copy of the string.
ASTERISK-21301 #close
Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06
During a transfer involving direct media a race occurs between when the
transferer channel is swapped out, initiating rtp changes/updates, and the
subsequent reinvites.
When Alice, after speaking with Charlie (Bob is on hold), connects Bob and
Charlie invites are sent to each in order to establish the call between them.
Bob is taken off hold and Charlie is told to have his media flow through
Asterisk. However, if before those invites go out the bridge updates Bob's
and/or Charlie's rtp information with direct media data (i.e. address, port)
then the invite(s) will contain the remote data in the SDP instead of the
Asterisk data.
The race occurs in the native bridge glue code when updating the peer. The
direct_media_address can get set twice before sending out the first invite
during call connection. This can happen because the checking/setting of the
direct_media_address happened in one thread while the sending of the invite(s)
happened in another thread.
This fix removes the race condition by moving the checking/setting of the
direct_media_address to be in the same thread as the sending of the invites(s).
This serializes the checking/setting and sending so they can no longer happen
out of order.
ASTERISK-25849 #close
Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
ASTERISK-25023 #close
Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
ASTERISK-25023
Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
* Made always run check_pendings() under the scheduler thread so scheduler
ids can be checked safely.
ASTERISK-25023
Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
ASTERISK-25023
Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
ASTERISK-25023
Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
ASTERISK-25023
Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
* Fix clearing autokillid in __sip_autodestruct() even though we could
reschedule.
ASTERISK-25023
Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
* Fix retrans_pkt() to call check_pendings() with both the owner channel
and the private objects locked as required.
* Refactor dialog retransmission packet list to safely remove packet
nodes. The list nodes are now ao2 objects. The list has a ref and the
scheduled entry has a ref.
ASTERISK-25023
Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641
This patch is part of a series to resolve deadlocks in chan_sip.c.
Stopping a scheduled event can result in a deadlock if the scheduled event
is running when you try to stop the event. If you hold a lock needed by
the scheduled event while trying to stop the scheduled event then a
deadlock can happen. The general strategy for resolving the deadlock
potential is to push the actual starting and stopping of the scheduled
events off onto the scheduler/do_monitor() thread by scheduling an
immediate one shot scheduled event. Some restructuring may be needed
because the code may assume that the start/stop of the scheduled events is
immediate.
ASTERISK-25023
Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48
This patch is part of a series to resolve deadlocks in chan_sip.c.
* Make dialog_unlink_all() unschedule all items at once in the sched
thread.
ASTERISK-25023
Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4
This patch is part of a series to resolve deadlocks in chan_sip.c.
The reordering of chan_sip's shutdown is to handle any immediate events
that get put onto the scheduler so resources aren't leaked. The typical
immediate events at this time are going to be concerned with stopping
other scheduled events.
ASTERISK-25023
Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20
This patch is part of a series to resolve deadlocks in chan_sip.c.
Delaying destruction of the chan_sip sip_pvt structures caused the
/channels/chan_sip/test_sip_rtpqos unit test to crash. That test
registers a special test ast_rtp_engine with the rtp engine module. When
the unit test completes it cleans up by unregistering the test
ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt
happens after the unit test returns, the destructor tries to call the rtp
engine destroy callback of the test ast_rtp_engine auto variable which no
longer exists on the stack.
* Change the test ast_rtp_engine auto variable to a static variable. Now
the variable can still exist after the unit test exits so the delayed
sip_pvt destruction can complete successfully.
ASTERISK-25023
Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13
Remove destructor calling destroy_it calling really_destroy_it
for no benefit. Just make the destructor the really_destroy_it
function.
Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a
Previous chan_sip behavior:
Before this patch chan_sip would always strip any quotes from an incoming
reason and pass that value up as the REDIRECTING(reason). For an outgoing
reason value, chan_sip would check the value against known values and
quote any it didn't recognize. Incoming 480 response message reason text
was just assigned to the REDIRECTING(reason).
Previous chan_pjsip behavior:
Before this patch chan_pjsip would always pass the incoming reason value
up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip
would send the reason value as passed down.
With this patch:
Both channel drivers match incoming reason values with values documented
by REDIRECTING(reason) and values documented by RFC5806 regardless of
whether they are quoted or not. RFC5806 values are mapped to the
equivalent REDIRECTING(reason) documented value and is set in
REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
quoted string version ('"unconditional"') is converted to
REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
with 'cfu' instead of any of the aliases.
The incoming 480 response reason text supported by chan_sip checks for
known reason values and if not matched then puts quotes around the reason
string and assigns that to REDIRECTING(reason).
Both channel drivers send outgoing known REDIRECTING(reason) values as the
unquoted RFC5806 equivalent. User custom values are either sent as is or
with added quotes if SIP doesn't allow a character within the value as
part of a RFC3261 Section 25.1 token. Note that there are still
limitations on what characters can be put in a custom user value. e.g.,
embedding quotes in the middle of the reason string is silly and just
going to cause you grief.
* Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
'cfu' value.
* Added missing malloc() NULL return check in res_pjsip_diversion.c
set_redirecting_reason().
* Fixed potential read from a stale pointer in res_pjsip_diversion.c
add_diversion_header(). The reason string needed to be copied into the
tdata memory pool to ensure that the string would always be available.
Otherwise, if the reason string returned by reason_code_to_str() was a
user's reason string then the string could be freed later by another
thread.
Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87
chan_sip could not handle AST_T38_TERMINATED frames being sent to it when
the channel left the bridge. The action resulted in overlapping outgoing
reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not
happy.
* Force T.38 to be remembered as locally bridged. Now when the channel
leaves the native RTP bridge after T.38, the channel remembers that it has
already reINVITEed the media back to Asterisk. It just needs to terminate
T.38 when the AST_T38_TERMINATED arrives.
* Prevent redundant AST_T38_TERMINATED from causing problems. Redundant
AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if
they happen before the T.38 state changes to disabled. Now the T.38 state
is set to disabled before the reINVITE is sent.
ASTERISK-25582 #close
Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce
Use the correct comparison function since we only care if the address
without the port is the same.
Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0
Previously you could add [!dnid] to the SIP dial string to alter the To:
header. This change allows you to alter the From header as well.
SIP dial string extra options now look like this:
[![touser[@todomain]][![fromuser][@fromdomain]]]
INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To:
header, that is no longer possible.
ASTERISK-25803 #close
Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7
Setting the sip.conf timert1 value to a value higher than 1245 can cause
an integer overflow and result in large retransmit timeout times. These
large timeout times hold system file descriptors hostage and can cause the
system to run out of file descriptors.
NOTE: The default sip.conf timert1 value is 500 which does not expose the
vulnerability.
* The overflow is now detected and the previous timeout time is
calculated.
ASTERISK-25397 #close
Reported by: Alexander Traud
Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a)
AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect
asterisk to include the same value for its own ip in both cases a) and b),
but it seems a) produces a contact header like Contact:
<sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like
<sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf
My guess is that manager_sipnotify should call
ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does,
because after applying this patch, both cases a) and b) produce
the contact header that I expect: <sip:asterisk@192.168.1.227:8060>
Reported by: Stefan Engström
Tested by: Stefan Engström
Change-Id: I86af5e209db64aab82c25417de6c768fb645f476
sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer
of 256 characters. This patch reduces the copy to 255 characters to leave
room for the string null terminator.
ASTERISK-25722 #close
Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab
In the sample sip.conf this is written with regard to notifyringing:
;notifyringing = no ; Control whether subscriptions already INUSE get sent
RINGING when another call is sent (default: yes)
However, this setting changes whether or not any RINGING indications are sent
to subscriptions. There is no separate configurable setting that allows
to control whether INUSE subscriptions also get sent RINGING. This is however
a useful option, to see (using BLF) if somebody else is able to handle an
incoming call or if everybody is busy.
This patch corrects the documentation for notifyringing (so the documentation
matches the functionality) and make notifyringing a tri-state option, by adding
the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing =
notinuse, only subscriptions that are not INUSE are sent the RINGING signal.
The default setting for notifyringing remains set to yes, so the default
behaviour is not affected.
ASTERISK-25558
Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa
websocket_write_timeout was not being set to its default value
during sip config reload, which meant that prior to this commit,
1) the default value of 100 was not used, unless an invalid value
(or 1) was specified in sip.conf for websocket_write_timeout, and
2) if the websocket_write_timeout directive was removed from sip.conf
without a full restart of asterisk, then the previous value would
continue to be used indefinitely.
This essentially lead to a 0ms write timeout (the first write attempt
in ast_careful_fwrite must have succeeded) in websocket write requests
from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf.
Changes to websocket_write_timeout still only apply to new websocket
sessions, after the sip reload -- timeouts on existing sessions are
not adjusted during sip reload.
Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953
Per the documentation the WebSocket support in chan_sip is
supposed to be enabled by default but is not. This change
corrects that.
Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423
Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously
this option was only being set on session sockets.
http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/
According to the link above, the SO_KEEPALIVE option is useful for knowing
when a TCP connected endpoint has severed communication without indicating
it or has become unreachable for some reason. Without this patch, keep
alive is not set on the socket listening for incoming TCP sessions and
in Komatsu's report this resulted in the thread listening for TCP becoming
stuck in a waiting state.
ASTERISK-25364 #close
Reported by: Hiroaki Komatsu
Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36
The current logic for ICE negotiation starts it
when receiving an SDP with ICE candidates. This is
incorrect as ICE negotiation can only start when each
call party have at least one pair of local and remote
candidate. Starting ICE negotiation early would result
in negotiation failure and ultimately no audio.
This change makes it so ICE negotiation is only started
when a response with SDP is received or when a response
with SDP is sent.
ASTERISK-24146
Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca
Asterisk may crash when calling ast_channel_get_t38_state(c)
on a locked channel which is being hung up.
ASTERISK-25609 #close
Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b
A crash happens sometimes when performing a CLI "sip reload". The bogus
peer gets refreshed while it is in use by a new call which can cause the
crash.
* Protected the global bogus peer object with an ao2 global object
container.
ASTERISK-25610 #close
Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed
Current support for reason header did work only in SIP responses.
According to RFC3336 the reason header might appear in any SIP request.
But it seems to make most sence in BYE and CANCEL so parasing is done
there too (if use_q850_reason=yes).
Change-Id: Ib6be7b34c23a76d0e98dfd0816c89931000ac790
zuul
Aaron An
Alexander Traud
George Joseph
Steve Davies
Walter Doekes
Joshua Colp
Kevin Harwell
Corey Farrell
cjack
Matt Jordan
Richard Mudgett
Mark Michelson
Timo Teräs
Vasil Kolev
kkm
Jaco Kroon
George Joseph
Francesco Castellano
Walter Doekes
StefanEng86
Ward van Wanrooij
Dade Brandon
Jonathan Rose
Eugene Voityuk
Filip Jenicek
Christof Lauber