If using a legitimate certificate from a trusted certificate authority,
you don't need to provide CA file.
Change-Id: I8623973b4209b44889243716d7880274caed8a6d
For some scenarios when an outgoing call was made only a subset of the
configured codecs were offered. If the codecs being offered happened to
not have a codec supported by the phone then the call would fail.
For instance Alice and Bob both are configured in Asterisk for g722 and ulaw(
allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's
only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with
Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains
g722 and not both g722 and ulaw, so the call ends.
This patch makes it so all the audio codecs configured on the endpoint always
get sent, and not just a subset. However priority is given to those codecs that
are compatible with the "other side".
ASTERISK-27259 #close
Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
When pruning a request to change the topology of a channel be
more intelligent about the resulting topology that is actually
used for SDP renegotiation.
In a case where a stream has not already been negotiated we
don't need to renegotiate and offer a declined stream. This can
occur if something in Asterisk (such as ConfBridge) requests
to add video to a PJSIP channel that has no video codecs configured.
In this case since the stream did not already exist we can safely
remove the stream from the requested topology, resulting in no
renegotiation occurring.
In a case where a renegotiation is requested with a codec that is
not supported we can reuse the formats of the existing stream if
it exists to ensure that the stream continues to flow, instead of
removing it.
Change-Id: I636540798d55922377318fe619c510fb6ed125fb
During a reinvite, if a remote endpoint error occurs and it returns a 500 the
session would end. This patch makes it so the session is not terminated, but
continues as it was.
The reason for this is because some endpoints may send non session terminating
"server errors" like a failed codec negotiation. So in this case instead of
ending the call it can hopefully continue. In the case of a real server error
the session is already "doomed", will be known soon enough and appropriately
ended by Asterisk later.
Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5
When an INVITE came in with both audio and video streams but there
were no audio codecs defined for the endpoint, we weren't declining
the audio stream. Since it's usually the first/transport stream,
when the video stream was processed and tried to use the transport,
it was empty and caused a crash. We now decline the the stream if
there are no matching codecs so when the video stream is processed,
it's now the first/transport stream and processes normally.
Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692
Assertions in the v15+ AST-2017-008 patches found that we were not
handling the case if the incoming SDP did not specify the required SSRC
attributes for bundled to work.
* Be strict on matching SSRC for bundled instances including the parent
instance. If the SSRC doesn't match then discard the packet. Bundled has
to tell us in the SDP signaling what SSRC to expect. Otherwise, we will
not know how to find the bundled instance structure.
Change-Id: I152830bbff71c662408909042068fada39e617f9
Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.
This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.
ASTERISK-27277
Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
There was an issue reported where an SDP received on a 183 Session
Progress message caused a crash because the pending streams had
already been processed when the OK was received. In that case the
pending topology was legitimately NULL. There was an assert for an
incorrect number of streams in the topology but not one for
topology being NULL. In any case, if you're not in dev-mode the
asserts don't do anything and since the scenario is legit, the
asserts weren't appropriate anyway.
* Changed several asserts to warning or debug messages and return
codes as appropriate.
ASTERISK-27264
Reported by: Daniel Heckl
Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848
In PostgreSQL 9.1 the backslash are string literals and not the escape
of characters.
In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the
support for old version of Postgresql than 9.1 was dropped. The sentence
before make was "ESCAPE '\'" but in version before than 9.1 need it to be
as follow "ESCAPE '\\'".
ASTERISK-27283
Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949
When a sip session is refreshed, the stream topology is looped
through, checking each stream for compatible formats. This would
cause a crash if the stream state was AST_STREAM_STATE_REMOVED,
since the formats would never be set for this stream, causing
a NULL value to be returned from ast_stream_get_formats. This
commit adds a check for streams with removed states.
Also removed a stray semicolon.
Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42
Previously, sRTP authentication failures were reported on log level WARNING.
When such failures happen, each RT(C)P packet is affected, spamming the log.
Now, those failures are reported at log level VERBOSE 2. Furthermore, the
amount is further reduced (previously all two seconds, now all three seconds).
Additionally, the new log entry informs whether media (RTP) or statistics (RTCP)
are affected.
ASTERISK-16898 #close
Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0
pubsub_on_rx_notify_request wasn't checking for a null
Content-Type header before checking that it was
application/simple-message-summary.
ASTERISK-27279
Reported by: Ross Beer
Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52
ast_variables_destroy is NULL safe, so there is no need to check its
argument before passing it.
ASTERISK-25524 #close
Reported by: Jesper
Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b
Validate RTCP packets before processing them.
* Validate that the received packet is of a minimum length and apply the
RFC3550 RTCP packet validation checks.
* Fixed potentially reading garbage beyond the received RTCP record data.
* Fixed rtp->themssrc only being set once when the remote could change
the SSRC. We would effectively stop handling the RTCP statistic records.
* Fixed rtp->themssrc to not treat a zero value as special by adding
rtp->themssrc_valid to indicate if rtp->themssrc is available.
ASTERISK-27274
Make strict RTP learning more flexible.
Direct media can cause strict RTP to attempt to learn a remote address
again before it has had a chance to learn the remote address the first
time. Because of the rapid relearn requests, strict RTP could latch onto
the first remote address and fail to latch onto the direct media remote
address. As a result, you have one way audio until the call is placed on
and off hold.
The new algorithm learns remote addresses for a set time (1.5 seconds)
before locking the remote address. In addition, we must see a configured
number of remote packets from the same address in a row before switching.
* Fixed strict RTP learning from always accepting the first new address
packet as the new stream.
* Fixed strict RTP to initialize the expected sequence number with the
last received sequence number instead of the last transmitted sequence
number.
* Fixed the predicted next sequence number calculation in
rtp_learning_rtp_seq_update() to handle overflow.
ASTERISK-27252
Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c
Incoming requests with non sip(s) URIs in the Request, To, From
or Contact URIs are now rejected with
PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in
pjsip_message_filter (formerly pjsip_message_ip_updater) and is
done at pjproject's "TRANSPORT" layer before a request can even
reach the distributor.
URIs read by res_pjsip_outbound_publish from pjsip.conf are now
also checked for both length and sip(s) scheme. Those URIs read
by outbound registration and aor were already being checked for
scheme but their error messages needed to be updated to include
scheme failure as well as length failure.
Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460
* The way that we were looking at XML elements for CalDAV was extremely
fragile, so use SAX2 for increased robustness.
* Don't complain about a 'channel' not be specified if autoreminder is
not set. Assume that if 'channel' is not set, we don't want to be
notified.
* Fix some truncated CLI output in 'calendar show calendar' and make the
'Autoreminder' description a bit more clear
ASTERISK-24588 #close
Reported by: Stefan Gofferje
ASTERISK-25523 #close
Reported by: Jesper
Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c
A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.
res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.
Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c
In 45744fc53, I mistakenly broke SDP media address rewriting by
misinterpreting which address was checked in the localnet comparison.
Instead of checking the remote peer address to decide whether we need
media address rewriting, we check our local media address: if it's
local, then we rewrite. This feels awkward, but works and even made
directmedia work properly if you set local_net. (For the record: for
local peers, the SDP media rewrite code is not called, so the
comparison does no harm there.)
ASTERISK-27248 #close
Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f
If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.
* control_swap_channel_in_bridge now only holds the control
lock while it's actually modifying the control structure and
releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.
Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3
Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro
SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with
previous versions of libSRTP, this change allows libSRTP 2.1.x as well.
ASTERISK-27253 #close
Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315
In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.
For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.
Therefore, checks like this look wrong, but are right:
/* See if where we are sending this request is local or not, and if
not that we can get a Contact URI to modify */
if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
ast_debug(5, "Request is being sent to local address, "
"skipping NAT manipulation\n");
(In the list == localnet == DENY == skip NAT manipulation.)
And conversely, other checks that looked right, were wrong.
This change adds two macro's to reduce the confusion and uses those
instead:
ast_sip_transport_is_nonlocal(transport_state, addr)
ast_sip_transport_is_local(transport_state, addr)
ASTERISK-27248 #close
Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934
Stream names within Asterisk can have meaning so when an externally
initiated renegotiation occurs we need to preserve the name of
the stream if it already exists.
Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596
t38_reinvite_response_cb can get called by res_pjsip_session's
session_inv_on_tsx_state_changed in situations where session->channel
is NULL. If it is, the ast_log warning segfaults because it tries
to get the channel name from a NULL channel.
* Check session->channel and print "unknown channel" when it's NULL.
ASTERISK-27236
Reported by: Ross Beer
Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7
sanitize_tdata was assuming all URIs were SIP URIs so when a non
SIP uri was in the From, To or Contact headers, the unconditional
cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused
a segfault when trying to access uri->other_param.
* Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri)
checks before attempting to cast or use the returned uri.
ASTERISK-27152
Reported-by: Ross Beer
Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received. The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.
* Add ast_safe_execvp() function. This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding. This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.
* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.
* Document code injection potential from untrusted data sources for other
shell commands that are under user control.
ASTERISK-27103
Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
This change moves the logic which learns a new source address
for RTP so it only occurs in the learning state. The learning
state is entered on initial allocation of RTP or if we are
told that the remote address for the media has changed. While
in the learning state if we continue to receive media from
the original source we restart the learning process. It is
only once we receive a sufficient number of RTP packets from
the new source that we will switch to it. Once this is done
the closed state is entered where all packets that do not
originate from the expected source are dropped.
The learning process has also been improved to take into
account the time between received packets so a flood of them
while in the learning state does not cause media to be switched.
Finally RTCP now drops packets which are not for the learned
SSRC if strict RTP is enabled.
ASTERISK-27013
Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c
When SDP renegotiation occurs it is possible for an RTP
instance to be reused for a new stream, resulting in the remote
SSRC changing if it is part of a bundle group. This change
allows this and updates its mapping in the current bundle
group.
ASTERISK-27231
Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490
If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.
ASTERISK-27209 #close
Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1
If a previously active stream is declined we could crash because the
channel's thread is still using the stream while we are updating the
topology in the serializer thread.
* Defer removing any declined stream's handler until we have blocked the
channel's thread with the channel lock.
ASTERISK-27212
Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420
fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon
success and -1 if the function is not available.
This commit inverts the return code check so that an error is printed if the
module is not loaded and not if it is loaded.
ASTERISK-27207 #close
Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb
When looking for recurring events, use the correct end time based on the
configured 'timeframe.'
ASTERISK-27174 #close
Reported by: Mark Thompson
Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef
* Check that the contact's reg_server matches the host's name before
deleting any prune_on_boot contacts. We don't want to delete reliable
transport contacts made with other servers if the ps_contacts database
table is shared with other servers.
Thanks to Ross Beer for pointing out that the original prune logic would
delete reliable transport contacts from other servers.
ASTERISK-27147
Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0
Add ability to use tokens instead of passwords according to Google OAuth 2.0
protocol.
ASTERISK-27169
Reported by: Andrey Egorov
Tested by: Andrey Egorov
Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db
The fix for the issue is broken up into three parts.
This is part three which handles the client side of REGISTER requests.
The registered contact may no longer be valid on the server when the
transport used is reliable and the connection is broken.
* Re-REGISTER our contact if the reliable transport is broken after
registration completes. We attempt to re-REGISTER immediately to minimize
the time we are unreachable. Time may have already passed between the
connection being broken and the loss being detected.
* Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's
are still correct if an allocation failure happens.
ASTERISK-27147
Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83
The fix for the issue is broken up into three parts.
This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled. Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.
* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown. If it is shutdown then the contact must be removed because it
is no longer valid. Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there. Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request. The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.
* Prune any rewrite_contact's registered reliable transport contacts on
boot. The reliable transport no longer exists so the contact is invalid.
* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.
* Made the websocket transport set a unique name since that is what we use
as the ao2 container key. Otherwise, we would not know which transport we
find when one of them shuts down. The names are also used for PJPROJECT
debug logging.
* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event. Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.
* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.
ASTERISK-27147
Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4
The fix for the issue is broken up into three parts.
This is part one which refactors the transport state monitor code to allow
more modules to be able to monitor transports.
* Pull the management of PJPROJECT's transport state callback code from
res_pjsip_transport_management.c into res_pjsip. Now other modules can
dynamically add and remove themselves from transport monitoring without
worrying about breaking PJPROJECT's callback chain.
* Add the ability for other modules to get a callback whenever a specific
transport is shutdown.
ASTERISK-27147
Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912
When handling an incoming SIP MESSAGE, PJSIP
attaches the IP address that the message was
received from to the message in the variable
PJSIP_RECVADDR. When the IP address is IPv6
the :PORT appended results in an unparseable
mess. By using an additional bit flag on the
pj_sockaddr_print call, the conventional use
of brackets around the address is achieved.
ASTERISK-27193 #close
Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9
Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated. Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.
ASTERISK-27158 #close
Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
ASTERISK-26745 #close
Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
A change was made long ago where the session was kept around
until the underlying INVITE session had been destroyed. This
had the side effect of also keeping the underlying media resources
around for this time as well.
This change ensures that when we are told to terminate the
session we immediately release any media sessions associated
with it.
ASTERISK-27110
Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82
Currently, the handling of the msid attribute is not quite right. According to
the spec the msid's between the offer/answer are not dependent upon one another.
Meaning the same msid's given in an offer do not have to be returned in the
answer for a given stream. And they probably shouldn't be (copied/reused) since
this can potentially cause some browser side confusion.
This patch generates new msids when both an offer and answer are sent from
Asterisk. However, Asterisk does reuse the original msid it sent out for a
reinvite. Also audio+video streams are paired together by sharing the same
stream id, but a different track id.
ASTERISK-27179 #close
Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643
* Remove unnecessary CMP_STOP.
* In handle_client_registration() use DEBUG_ATLEAST() to only do work
needed for the debug log message when the debug log message is needed.
* In sip_outbound_registration_state_destroy() check state->registration
for NULL.
Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80
Most uses of CMP_STOP are superfluous and are only respected when
OBJ_MULTIPLE is used to search the container.
Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8
Support building the Asterisk httpd with version 3.0 of gmime as
well as earlier versions of that library.
ASTERISK-27173
Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f
When the "webrtc" option was added in res_pjsip it was not added to the alembic
scripts. This patch adds the option for alembic.
Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of
an OPT_BOOL_T so if this field is ever written to a database it will write out
the correct value.
ASTERISK-27119 #close
Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b
This change fixes PIDF content generation when the underlying device
state is considered in use. Previously it was incorrectly marked
as closed meaning they were offline/unavailable. The code now
correctly marks them as open.
Additionally:
* Generate an XML element for our activity instead of a using a text
node.
* Consider every extension state other than "unavailable" to be 'open'
status.
* Update the XML namespaces and structure to reflect those
documented in RFC 4480
* Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the
"in use" activity. This change results in eyeBeam using the
appropriate icon for the watched user.
This was tested on eyeBeam 1.5.20.2 build 59030 on Windows.
ASTERISK-26659 #close
Reported by: Abraham Liebsch
patches:
ASTERISK-26659.diff submitted by snuffy (license 5024)
Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810
The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.
Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
Case scenario with Applications ARI:
* Once you subscribe to deviceState with Applications REST API, it will be
added into subscription pool.
* When you unsubscribe it will remove from the device_state_subscription
hash table but not from the subscription pool.
* When you subscribe again, it will add it to pool again.
* Now you will have two subscriptions and you will receive same event
twice.
This fix should now remove deviceState subscription from pool and it
should fix unsubscribe on deviceState.
ASTERISK-27130 #close
Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4
This change does a few things to improve packet loss and renegotiation:
1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.
2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.
3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.
4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.
5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.
ASTERISK-27143
Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
This change makes it so that if an RTCP packet is being sent
the RTP ICE component is used for sending if RTCP-MUX is in use.
ASTERISK-27133
Change-Id: I6200f611ede709602ee9b89501720c29545ed68b
This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:
rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.
Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.
ASTERISK-27119 #close
Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.
This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.
For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.
ASTERISK-27118
Change-Id: I96c0920b9f9aca7382256484765a239017973c11
Currently when rtp is paused, no packets are written to the
recorded audio file, causing the silence to be skipped and recording
not properly time aligned. The read handler as been adapted to
return a silence frame of the correct size.
ASTERISK-27128 #close
Change-Id: I2d7f60650457860b9c70907b14426756b058a844
arm the t38 webhook always, so we can correctly reject a
T38 negotiation request when t38 is disabled on a channel
Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d
This adds a parameter to ast_waitfordigit_full which can be used to only
stop waiting when certain expected digits are received. Any unexpected
DTMF digits are simply ignored.
This also creates a new dialplan application WaitDigit.
ASTERISK-27129 #close
Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9
By default, when res_musiconhold reloads or unloads, it sends a HUP
signal to custom applications (and all descendants), waits 100ms,
then sends a TERM signal, waits 100ms, then finally sends a KILL
signal. An application which is interacting with an external
device and/or spawns children of its own may not be able to exit
cleanly in the default times, expecially if sent a KILL signal, or
if it's children are getting signals directly from
res_musiconhoild.
* To allow extra time, the 'kill_escalation_delay'
class option can be used to set the number of milliseconds
res_musiconhold waits before escalating kill signals, with the
default being the current 100ms.
* To control to whom the signals are sent, the "kill_method" class
option can be set to "process_group" (the default, existing
behavior), which sends signals to the application and its
descendants directly, or "process" which sends signals only to the
application itself.
Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b
If the from_user field contains certain characters (like @, {, ^, etc.),
PJSIP will return a null value for the URI when attempting to parse it.
This causes a crash when trying to dial out through a trunk that contains
these invalid characters in its from_user field.
This change checks the configuration and ensures that an endpoint will
not be created if the from_user contains an invalid character. It also
adds a null check to the PJSIP URI parsing as a backup.
ASTERISK-27036 #close
Reported by: Maxim Vasilev
Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0
When a message is received on the TURN socket, the code processing the
message needs to call into the ICE/STUN session for further processing.
This code path locks the TURN group lock then the ICE/STUN group lock. In
another thread an ICE/STUN timer can fire off to send a keep alive message
over the TURN socket. In this code path, the ICE/STUN group lock is
obtained then the TURN group lock is obtained to send the packet. A
classic deadlock case if the group locks are not the same.
* Made TURN get created using the ICE/STUN session's group lock.
NOTE: I was originally concerned that the ICE/STUN session can get
recreated by ice_reset_session() for an event like RTCP multiplexing
causing a change during SDP negotiation. In this case the TURN group lock
would become different. However, TURN is also recreated as part of the
ICE/STUN recreation in ice_create() when all known ICE candidates are
added to the new ICE session. While the ICE/STUN and TURN sessions are
being recreated there is a period where the group locks could be
different.
ASTERISK-27023 #close
Patches:
res_rtp_asterisk-turn-deadlock-fix.patch (license #6502)
patch uploaded by Michael Walton (modified)
Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9
This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.
Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83
When a SIP message comes in on a transport, pjproject obtains the lock on
the transport and pulls the data out of the socket. Unlike UDP, the TCP
transport does not allow concurrent access. Without concurrency the
transport lock is not released when the transport's message complete
callback is called. The processing continues and eventually Asterisk
starts processing the SIP message. The first thing Asterisk tries to do
is determine the associated dialog of the message to determine the
associated serializer. To get the associated serializer safely requires
us to get the dialog lock.
To send a request or response message for a dialog, pjproject obtains the
dialog lock and then obtains the transport lock. Deadlock can result
because of the opposite order the locks are obtained.
* Fix the deadlock by obtaining the serializer associated with the dialog
another way that doesn't involve obtaining the dialog lock. In this case,
we use an ao2 container to hold the associated endpoint and serializer.
The new locks are held a brief time and won't overlap other existing lock
times.
ASTERISK-27090 #close
Change-Id: I9ed63f4da9649e9db6ed4be29c360968917a89bd
The OBJ_SEARCH_xxx defines should not be used as if they were individual
bits. They represent a multi-bit enumeration value field.
Change-Id: I32abc9a475396dab02402a7014357dd94284e17b
There needed to be a way to notify handlers upstream that DTLS had been
established. This patch makes it so once DTLS has been estalished a source
change control frame is put into the read queue. Any handlers can then watch
for that frame and trigger off of it.
ASTERISK-27096 #close
Change-Id: I27ff344f5a8c691a1890dfe3254a4b1a49e7f4a0
When connected_line_method is "invite", we're supposed to determine
if the client can support UPDATE and if it can, send UPDATE instead
of INVITE to avoid the SDP renegotiation. Not only was pjproject
not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing
that invite_tsx wasn't NULL which isn't always the case.
* Updated chan_pjsip/update_connected_line_information to drop the
requirement that invite_tsx isn't NULL.
* Submitted patch to pjproject sip_inv.c that sets the
PJSIP_INV_SUPPORT_UPDATE flag correctly.
* Updated pjsip.conf.sample to clarify what happens when "invite"
is specified.
ASTERISK-27095
Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560
The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated. This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.
ASTERISK-27066 #close
Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
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
When re-inviting to add more streams it is possible for
the role of existing ICE sessions to be changed to the
incorrect value. This results in subsequent refreshes
within the sessions getting a role conflict and the ICE
session breaking down. This change only sets the role to
be the new value if an ICE renegotiation is actually
going to happen, otherwise the existing role is preserved.
As well if we encounter a situation where a unidirectional
ICE negotiation happens and the other side does not send us
candidates we will not store any information for sending
traffic, even though we know where they are reachable. This
change fixes this by using the source of the ICE traffic
itself as the target if no candidates are known and we
receive some ICE traffic.
ASTERISK-27088
Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9
The T38 sdp callback incorrectly has a side effect of incrementing
the media_count. This can lead to core dumps.
Change-Id: I7bb2f4987de4046ec52cfc34e5ea0662dae32af8
Do not need to unsubscribe/subscribe on creating the ednpoint's contact.
The modified function create_mwi_subscriptions_for_endpoint adds
the subscription only if it does not exist.
The subscriptions aren't added for active contacts
which are retrieved on startup from realtime
if mwi_disable_initial_unsolicited=yes.
Because the mwi_contact_added is not called.
So the subscriptions also should be created on updating contact.
ASTERISK-26230 #close
Change-Id: I47e265af9296ca09aa42a316fdacac104148cee4
The SDP offer/answer model requires an answer to an offer before a new SDP
can be processed. This allows our local SDP creation to be deferred until
we know that we need to create an offer or an answer SDP. Once the local
SDP is created it won't change until the SDP negotiation is restarted.
An offer SDP in an initial SIP INVITE can receive more than one answer
SDP. In this case, we need to merge each answer SDP with our original
offer capabilities to get the currently negotiated capabilities. To
satisfy this requirement means that we cannot update our proposed
capabilities until the negotiations are restarted.
Local topology updates from ast_sdp_state_update_local_topology() are
merged together until the next offer SDP is created. These accumulated
updates are then merged with the current negotiated capabilities to create
the new proposed capabilities that the offer SDP is built.
Local topology updates are merged in several passes to attempt to be smart
about how streams from the system are matched with the previously
negotiated stream slots. To allow for T.38 support when merging, type
matching considers audio and image types to be equivalent. First streams
are matched by stream name and type. Then streams are matched by stream
type only. Any remaining unmatched existing streams are declined. Any
new active streams are either backfilled into pre-merge declined slots or
appended onto the end of the merged topology. Any excess new streams
above the maximum supported number of streams are simply discarded.
Remote topology negotiation merges depend if the topology is an offer or
answer. An offer remote topology negotiation dictates the stream slot
ordering and new streams can be added. A remote offer can do anything to
the previously negotiated streams except reduce the number of stream
slots. An answer remote topology negotiation is limited to what our offer
requested. The answer can only decline streams, pick codecs from the
offered list, or indicate the remote's stream hold state.
I had originally kept the RTP instance if the remote offer SDP changed a
stream type between audio and video since they both use RTP. However, I
later removed this support in favor of simply creating a new RTP instance
since the stream's purpose has to be changing anyway. Any RTP packets
from the old stream type might cause mischief for the bridged peer.
* Added ast_sdp_state_restart_negotiations() to restart the SDP
offer/answer negotiations. We will thus know to create a new local SDP
when it is time to create an offer or answer.
* Removed ast_sdp_state_reset(). Save the current topology before
starting T.38. To recover from T.38 simply update the local topology to
the saved topology and restart the SDP negotiations to get the offer SDP
renegotiating the previous configuration.
* Allow initial topology for ast_sdp_state_alloc() to be NULL so an
initial remote offer SDP can dictate the streams we start with. We can
always update the local topology later if it turns out we need to offer
SDP first because the remote chose to defer sending us a SDP.
* Made the ast_sdp_state_alloc() initial topology limit to max_streams,
limit to configured codecs, handle declined streams, and discard
unsupported types.
* Convert struct ast_sdp to ao2 object. Needed to easily save off a
remote SDP to refer to later for various reasons such as generating
declined m= lines in the local SDP.
* Improve converting remote SDP streams to a topology including stream
state. A stream state of AST_STREAM_STATE_REMOVED indicates the stream is
declined/dead.
* Improve merging streams to take into account the stream state.
* Added query for remote hold state.
* Added maximum streams allowed SDP config option.
* Added ability to create new streams as needed. New streams are created
with configured default audio, video, or image codecs depending on stream
type.
* Added global locally_held state along with a per stream local hold
state. Historically, Asterisk only has a global locally held state
because when the we put the remote on hold we do it for all active
streams.
* Added queries for a rejected offer and current SDP negotiation role.
The rejected query allows the using module to know how to respond to a
failed remote SDP set. Should the using module respond with a 488 Not
Acceptable Here or 500 Internal Error to the offer SDP?
* Moved sdp_state_capabilities.connection_address to ast_sdp_state. There
seems no reason to keep it in the sdp_state_capabilities struct since it
was only used by the ast_sdp_state.proposed_capabilities instance.
* Callbacks are now available to allow the using module some customization
of negotiated streams and to complete setting up streams for use. See the
typedef doxygen for each callback for what is allowable and when they are
called.
* Added topology answerer modify callback.
* Added topology pre and post apply callbacks.
* Added topology offerer modify callback.
* Added topology offerer configure callback.
* Had to rework the unit tests because I changed how SDP topologies are
merged. Replaced several unit tests with new negotiation tests.
Change-Id: If07fe6d79fbdce33968a9401d41d908385043a06
If the endpoint's last contact is deleted unsolicited MWI has to be
unsubscribed.
ASTERISK-27051 #close
Change-Id: I33e174e0b9dba0998927d16d6d100fda5c7254e0
When a stasis channel is stolen by another app, the control
structure is unreffed but never unlinked from the app_controls
container. This causes the channel reference to leak.
Added OBJ_UNLINK to the callback in channel_stolen_cb.
Also added some additional channel lifecycle debug messages to
channel.c.
ASTERISK-27059 #close
Repoorted-by: George Joseph
Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14
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
In Corosync 2.x libraries were changed to use LibQB IPC.
Sadly LibQB IPC doesn't support copy-free access to received buffer, so
Corosync libraries were rewritten to use stack as buffer. Mostly the
needed stack size is quite small, but for all *_dispatch functions, 1MiB
is needed.
Asterisk function ast_pthread_create_background set stack size for new
thread to much smaller AST_BACKGROUND_STACKSIZE (~500KiB).
This results in Asterisk crash when running with Corosync 2.x.
Patch solves this issue by creating it's own version of
ast_pthread_create_background which sets stack size to much higher value
(actually it's AST_BACKGROUND_STACKSIZE + 3MiB).
Another problem may appear when "corosync show members" netconsole
command is executed. It is also executed in thread and also has only
500KiB stack size. Sadly it calls corosync_cfg_get_node_addrs which
again needs at least 1MiB stack.
Solution is to use HAVE_COROSYNC_CFG_STATE_TRACK as a discriminator
between Corosync 1.x and 2.x. If 1.x is found, nothing changes. If 2.x
is found, NodeID is displayed instead of IP address.
ASTERISK-25370 #close
Reported by: mdu113
Change-Id: Id95b0d21ab6e708e7d74ad8786c587211676fa08
The recent change to make the use of LOAD_DECLINE more consistent
caused res_ari to unload itself before declining if the ari.conf
file wasn't found. The ari stubs though still tried to use the
configuration resulting in segfaults.
This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests
to see if res_ari is actually loaded and causes the stubs to also
decline if it isn't. The macro was then added to the mustache
template's "load_module" function.
ASTERISK-27026 #close
Reported-by: Ronald Raikes
Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d
destroy_subscription was attempting to get the id of the
subscription tree's endpoint after we'd already called ao2_cleanup
on it causing a segfault.
Moved the cleanup until after the debug statement and since
endpoint could also be NULL at this point, check for that as well.
ASTERISK-27057 #close
Reported-by: Ryan Smith
Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678
There was a typo introduced in commit 776ffd77 which was preventing
the transport's external media address from being used.
ASTERISK-27024 #close
Reported-by: Christopher van de Sande
patches:
patch.diff submitted by Florian Floimair (license 6892)
Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27
It looks like there was a copy/paste error in ast_rtp_change_source
where if there was a rtcp srtp instance, instead of updating its
ssrc we were updating the srtp instance ssrc twice.
ASTERISK-27022 #close
Reported-by: Michael Walton
Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095
When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.
This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.
ASTERISK-27053 #close
Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
If sending unsolicited mwi to all endpoints on startup is disabled
(mwi_disable_initial_unsolicited=yes) do not need to create subscriptions.
If there are many (thousands) realtime endpoints configured with unsolicited mwi
and Vociemail Storage configured as ODBC or IMAP there will be huge number of
DB/IMAP requests on startup.
ASTERISK-26230 #close
Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5
Added check for NULL return value when calling
ast_sorcery_retrieve_by_id in function get_write_timeout
ASTERISK-27046
Change-Id: I9357717278da631c3a1cb502c412693929b0cb41
PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.
This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.
ASTERISK-26996
Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
When a frame destined for a MulticastRTP channel does not have timing
information (such as when an 'originate' is done), we generate the RTP
timestamps ourselves without regard to the number of samples we are
about to send.
Instead, use the same method as res_rtp_asterisk and 'predict' a
timestamp given the number of samples. If the difference between the
timestamp that we generate and the one we predict is within a specific
threshold, use the predicted timestamp so that we end up with timestamps
that are consistent with the number of samples we are actually sending.
Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f
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
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
This change allows the format of the EAGI audio pipe to be changed by
setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of
the loaded formats.
ASTERISK-26124 #close
Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd
Documented the 'beep' option in both the parameters list and the command
description.
ASTERISK-23839 #close
Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea
Explicitly check that the appropriate number of arguments were passed to
SET VARIABLE before attempting to reference them. Also initialize the
arguments array to zeroes before populating it.
ASTERISK-22432 #close
Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97
If the generated XML documentation for a command does not end with a \n,
the postamble of the usage message does not appear on its own line.
ASTERISK-25662 #close
Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020
Some devices separate format attributes with a semicolon followed by a
space, so trim blanks before trying to match them.
ASTERISK-27008 #close
Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc
When using rtcp mux if an rtcp payload came in it would still use the srtp
unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp
data was being passed to the rtp unprotect method this would result in an
error.
This patch ensures that the correct unprotect method is chosen by making
sure the passed in rtcp flag is appropriately set when rtcp mux is enabled
and an rtcp payload is received.
ASTERISK-26979 #close
Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241
When manipulating flags on a channel the channel has to be
locked to guarantee that nothing else is also manipulating
the flags. This change introduces locking where necessary to
guarantee this. It also adds helper functions that manipulate
channel flags and lock to reduce repeated code.
ASTERISK-26789
Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10
Retransmissions of an initial INVITE could be queued in the serializer
before we have processed the first INVITE message. If the first INVITE
message doesn't get completely processed before the retransmissions are
seen then we could try to setup the same call from the retransmissions. A
symptom of this is seeing a (key exists) message associated with an
INVITE. An earlier change attempted to address this kind of problem by
calculating a distributor serializer to use for unassociated messages.
Part of that change also made incoming calls keep using that distributor
serializer. (ASTERISK-26088) However, some leftover code was still
deferring the INVITE processing to the session's serializer even though we
were already in that serializer. This not only is unnecessary but would
cause the same call resetup problem.
* Removed the code to defer processing the initial INVITE to the session's
serializer because we are already running in that serializer.
ASTERISK-26998 #close
Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6
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
This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.
Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support
ASTERISK-26427
Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
When a call gets put on hold RTP is temporarily stopped and Asterisk was
setting the remote RTCP address to NULL. Then when RTCP data was received
from the remote endpoint, Asterisk would be missing this information when
publishing the rtcp_message stasis event. Consequently, message subscribers
(in this case res_hep_rtcp) trying to parse the "from" field output the
following error:
"ast_sockaddr_split_hostport: Port missing in (null)"
This patch makes it so the remote RTCP address is no longer set to NULL when
stopping RTP. There was only one place that appeared to check if the remote
RTCP address was NULL as a way to tell if RTCP was running. This patch added
an additional check on the RTCP schedid for that case to make sure RTCP was
truly not running.
ASTERISK-26860 #close
Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b
A deadlock can happen between a channel lock and a pjsip session media
container lock. One thread is processing a reINVITE's SDP and walking
through the session's media container when it waits for the channel lock
to put the determined format capabilities onto the channel. The other
thread is writing a frame to the channel and processing the T.38 frame
hook. The T.38 frame hook then waits for the pjsip session's media
container lock. The two threads are now deadlocked.
* Made the T.38 frame hook release the channel lock before searching the
session's media container. This fix has been done to several other
frame hooks to fix deadlocks.
ASTERISK-26974 #close
Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186
There was no context info in this module's log messages so it was
impossible to toubleshoot.
Added endpoint or host to all messages and added the realms in the
challenge for the "No auth credentials for any realm" message.
Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b
RFC 5576 defines how SSRC-level attributes may be added to SDP media
descriptions. In general, this is useful for grouping related SSRCes,
indicating SSRC-level format attributes, and resolving collisions in RTP
SSRC values. These attributes are used widely by browsers during WebRTC
communications, including attributes defined by documents outside of RFC
5576.
This commit introduces the addition of SSRC-level attributes into SDPs
generated by Asterisk. Since Asterisk does not tend to use multiple
SSRCs on a media stream, the initial support is minimal. Asterisk
includes an SSRC-level CNAME attribute if configured to do so. This at
least gives browsers (and possibly others) the ability to resolve SSRC
collisions at offer-answer time.
In order to facilitate this, the RTP engine API has been enhanced to be
able to retrieve the SSRC and CNAME on a given RTP instance.
res_rtp_asterisk currently does not provide meaningful CNAME values in
its RTCP SDES items, and therefore it currently will always return an
empty string as the CNAME value. A task in the near future will result
in res_rtp_asterisk generating more meaningful CNAMEs.
Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789
If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
Joshua Colp
Sean Bright
Kevin Harwell
George Joseph
Richard Mudgett
Jenkins2
Rodrigo Ramírez Norambuena
Ben Ford
Alexander Traud
Walter Doekes
Corey Farrell
Torrey Searle
Michael Kuron
Andrey Egorov
Scott Griepentrog
Tzafrir Cohen
Sergej Kasumovic
Mark Michelson
Alexei Gradinari
Jan Friesse
Jørgen H
Martin Tomec