Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.
This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.
#SIPit31
ASTERISK-24382 #close
Reported by: Matt Jordan
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Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
ASTERISK-24356 #close
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/4034/
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This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.
Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.
Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.
ASTERISK-24348 #related
Review: https://reviewboard.asterisk.org/r/4015/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Outgoing PJSIP calls can result in non-negotiated formats listed in the
channel's native formats if video formats are listed in the endpoint's
configuration. The resulting call could then use a non-negotiated format
resulting in one way audio.
* Simplified the update of session->req_caps in set_caps(). Why do
something in five steps when only one is needed?
AFS-162 #close
Review: https://reviewboard.asterisk.org/r/4000/
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This module supports sending both unicast and multicast RTP
to a specified target. Multicast functionality is the same as
chan_multicast_rtp was. In the case of unicast a specific
IP address and port can be specified, along with optional RTP
engine and format in the form of:
UnicastRTP/<ip address>:<port>/<engine>/<format>
This can be useful for sending a copy of a media stream to
another application for processing.
Review: https://reviewboard.asterisk.org/r/3981/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Testsuite tests will occasionally fail because on reception of a 200 OK SIP response,
an AST_CONTROL_ANSWER frame is queued prior to when media has finished being
negotiated. This is because session supplements are called into before PJSIP's
inv_session code has told us that media has been updated. Sometimes the queued answer
frame is handled by the PBX thread before the ensuing media negotiations occur, causing
a test failure.
As it turns out, there is another place that session supplements could be called into, which is
after media has finished getting negotiated. What this commit introduces is a means for session
supplements to indicate when they wish to be called into when handling an incoming SIP response.
By default, all session supplements will be run at the same point that they were prior to this
commit. However, session supplements may indicate that they wish to be handled earlier than
normal on redirects, or they may indicate they wish to be handled after media has been negotiated.
In this changeset, two session supplements have been updated to indicate a preference for when
they should be run: res_pjsip_diversion executes before handling redirection in order to get
information from the Diversion header, and chan_pjsip now handles responses to INVITEs after
media negotiation to fix the race condition mentioned previously.
ASTERISK-24212 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/3930
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The reporter on the issue found some issues when upgrading from version 10 to 11
on 55 hosts.
Two situations that can occur with dynamic registrations.
1. With dnsmgr disabled, if the host is not resolvable we are not trying to
resolve the host again when it is time to attempt to register again. This
results in never registering to the host.
2. With dnsmgr enabled, when the host is temporarily not resolvable the
address is set to 0.0.0.0:0 and then when the host is resolvable the port
is not being restored and stays set to 0.
This patch resolves these two issues by:
* Storing the hostname so that it can be used for resolving with DNS.
* Resolve the hostname on the next scheduled attempt to register.
* Storing the port used to reach the host so that when the hostname is
resolvable again, we can set the port again if the port is still unset after
looking up the host.
ASTERISK-23767 #close
Reported by: David Herselman
Tested by: David Herselman, Michael L. Young
Patches:
asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3856/
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Users now have the ability to bind the rtpengine instance to a specific IP
address. For example, you want chan_sip (call control) on eth0 but rtp (media)
on eth1.
ASTERISK-24280 #close
Reported by: Paul Belanger
Tested by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/3952/
Patches:
rtpengine.diff uploaded by Paul Belanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code originally worked around an issue within res_rtp_asterisk itself.
The wrong socket was being used for the STUN check for RTCP, causing the
port to be the same as RTP. This was subsequently fixed and the RTCP port
provided for the ICE candidate is correct and does not need to be incremented.
ASTERISK-23997 #close
Reported by: Badalian Vyacheslav
Patches:
plus1.diff submitted by Badalian Vyacheslav (license 5249)
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On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.
* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite. AFS-63 was effectively reintroduced because of the media
formats work. res_pjsip_sdp_rtp.c:set_caps()
* Improved the unexpected frame format WARNING message to include more
information.
* Added protective locking while altering formats on a channel. Reworked
set_format() to simplify and protect the formats under manipulation.
* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())
AFS-137 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3906/
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A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
while C has the full information about A
I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:
* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id. This is why party A got
default connected line information.
* Made update_initial_connected_line() use the channel's CALLERID(id)
information. The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.
* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id. This includes the configured
callerid_tag string and other party id fields.
* Fixed accessing channel party id information without the channel lock
held.
* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock. Shallow copy string pointers can
become stale if the channel lock is not held.
* Made queue_connected_line_update() also update the channel's
CALLERID(id) information. Moving the channel to another bridge would need
the information there for the new bridge peer.
* Fixed off nominal memory leak in update_incoming_connected_line().
* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().
AFS-98 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3913/
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* Fixed the iax.conf bandwidth option. This is the root cause of
ASTERISK-24150.
* Added checks in iax2_request() to ensure that there are actual formats
requested for the new channel to prevent any more fracks from issues like
ASTERISK-24150. This is a consequence of the iax.conf bandwidth option
not working.
* Fixed struct iax2_codec_pref.order member size mismatch issue when
converting to and from the codec preference order list passed over the
wire. In addition the values sent over the wire are now compatible with
previous Asterisk versions.
* Fixed several issues dealing with the struct iax2_codec_pref members.
Off-by-one, array limit errors, and the order/framing members always need
to be updated together.
* Made iax2_request() setup the channel's native format preference order
according to the user's wishes. The new media format strategy needs the
order specified earler.
* Fixed usage of ast_format_compatibility_bitfield2format(). The function
can return NULL if the bitfield was not associated with a function.
* Deleted dead code iax2_codec_pref_getsize() and
iax2_codec_pref_setsize().
* Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call
iax2_codec_pref_to_cap() instead of inlining it.
* Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and
IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8.
* Renamed prefs to prefs_global so it won't get confused with the local
pref versions.
* Fixed too small buffer in handle_cli_iax2_show_peer().
* Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete
lines.
* Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an
optimization so iax2_request() and iax2_call() do less work.
* Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when
the pbx could not get started.
* Made set_config() setup a local prefs list along side the local
capability format bitfield. Once the config is loaded, then the local
copies are put into the global versions.
* Fix unininialized codec_buf in function_iaxpeer().
ASTERISK-24150 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3890/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
ARI: Add channel technology agnostic out of call text messaging
This patch adds the ability to send and receive text messages from various
technology stacks in Asterisk through ARI. This includes chan_sip (sip),
res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the
endpoints resource, and can be sent directly through that resource, or to a
particular endpoint.
For example, the following would send the message "Hello there" to PJSIP
endpoint alice with a display URI of sip:asterisk@mycooldomain.org:
ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
This is equivalent to the following as well:
ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
Both forms are available for message technologies that allow for arbitrary
destinations, such as chan_sip.
Inbound messages can now be received over ARI as well. An ARI application that
subscribes to endpoints will receive messages from those endpoints:
{
"type": "TextMessageReceived",
"timestamp": "2014-07-12T22:53:13.494-0500",
"endpoint": {
"technology": "PJSIP",
"resource": "alice",
"state": "online",
"channel_ids": []
},
"message": {
"from": "\"alice\" <sip:alice@127.0.0.1>",
"to": "pjsip:asterisk@127.0.0.1",
"body": "Watson, come here.",
"variables": []
},
"application": "testsuite"
}
The above was made possible due to some rather major changes in the message
core. This includes (but is not limited to):
- Users of the message API can now register message handlers. A handler has
two callbacks: one to determine if the handler has a destination for the
message, and another to handle it.
- All dialplan functionality of handling a message was moved into a message
handler provided by the message API.
- Messages can now have the technology/endpoint associated with them.
Various other properties are also now more easily accessible.
- A number of ao2 containers that weren't really needed were replaced with
vectors. Iteration over ao2_containers is expensive and pointless when
the lifetime of things is well defined and the number of things is very
small.
res_stasis now has a new file that makes up its structure, messaging. The
messaging functionality implements a message handler, and passes received
messages that match an interested endpoint over to the app for processing.
Note that inadvertently while testing this, I reproduced ASTERISK-23969.
res_pjsip_messaging was incorrectly parsing out the 'to' field, such that
arbitrary SIP URIs mangled the endpoint lookup. This patch includes the
fix for that as well.
Review: https://reviewboard.asterisk.org/r/3726
ASTERISK-23692 #close
Reported by: Matt Jordan
ASTERISK-23969 #close
Reported by: Andrew Nagy
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r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Remove automerge properties :-(
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r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
test_message: Fix strict-aliasing compilation issue
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This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes
to the RTCP topics in Stasis and receives RTCP information back from the
message bus. It encodes into HEPv3 packets and sends the information to the
res_hep module for transmission.
Using this, someone with a Homer server can get live call quality monitoring
for all RTP-based channels in their Asterisk 12+ systems.
In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and
chan_pjsip that were uncovered by the tests written for the Asterisk Test
Suite. This patch fixes the following:
1) chan_pjsip failed to set its channel unique ids on its RTP instance on
outbound calls. It now does this in the appropriate location, in the
serialized call callback.
2) The rtp_engine was overflowing some values when packed into JSON.
Specifically, some longs and unsigned ints can't be be packed into integer
values, for obvious reasons. Since libjansson only supports integers,
floats, strings, booleans, and objects, we print these values into strings.
3) res_rtp_asterisk had a few problems:
(a) it would emit a source IP address of 0.0.0.0 if bound to that IP
address. We now use ast_find_ourip to get a better IP address, and
properly marshal the result into an ast_strdupa'd string.
(b) Reports can be generated with no report bodies. In particular, this
occurs when a sender is transmitting information to a receiver (who
will send no RTP back to the sender). As such, the sender has no report
body for what it received. We now properly handle this case, and the
sender will emit SR reports with no body. Likewise, if we receive an
RTCP packet with no report body, we will still generate the appropriate
events.
ASTERISK-24119 #close
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In Asterisk, it is possible for a device to have a status of ONHOLD. This is
not typically an easy thing to determine, as a channel being on hold is not
a direct channel state. Typically, this has to be calculated outside of the
core independently in channel drivers, notably, chan_sip and chan_pjsip. Both
of these channel drivers already have to calculate device state in a fashion
more complex than the core can handle, as they aggregate all state of all
channels associated with a peer/endpoint; they also independently track
whether or not one of those channels is currently on hold and mark the device
state appropriately.
In 12+, we now have the ability to report an AST_DEVICE_ONHOLD state for all
channels that defer their device state to the core. This is due to channel hold
state actually now being tracked on the channel itself. If a channel driver
defers its device state to the core (which many, such as DAHDI, IAX2, and
others do in most situations), the device state core already goes out to get a
channel associated with the device. As such, it can now also factor the channel
hold state in its calculation.
This patch adds this logic to the device state core. It also uses an existing
mapping between device state and channel state to handle more channel states.
chan_pjsip has been updated slightly as well to make use of this (as it was,
for some reason, reporting a channel state of BUSY as a device state of INUSE,
which feels slightly wrong).
Review: https://reviewboard.asterisk.org/r/3771/
ASTERISK-24038 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
for sending/receiving arbitrary out of call text messages through ARI in a
technology agnostic fashion.
The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
arbitrary technology defined URI. This is less straight forward, as
endpoints are formed from a tech + resource pair. We don't have a
mechanism to note that a technology that *may* have endpoints exists.
This patch provides such a mechanism, and fixes a few bugs along the way.
The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
most of the interesting bits (such as channel creation, destruction, state
changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
This resulted in endpoints missing the channel creation message, which
limited the usefulness of the subscription in the first place (a major use
case being 'tell me when this endpoint has a channel'). Unfortunately,
this meant another parameter to ast_channel_alloc. Since not all channel
technologies support an ast_endpoint, this patch makes such a call
optional and opts for a new function, ast_channel_alloc_with_endpoint.
When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.
Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:
channel PJSIP/foo-1 --
\
--> endpoint PJSIP/foo --
/ \
channel PJSIP/foo-2 -- \
---- > endpoint PJSIP
/
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --
ARI, through the applications resource, can:
- subscribe to endpoint:PJSIP/foo and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
- subscribe to endpoint:PJSIP/bar and get notifications for channels
PJSIP/bar-1 and endpoint PJSIP/bar
- subscribe to endpoint:PJSIP and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar
Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).
This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).
Review: https://reviewboard.asterisk.org/r/3760/
ASTERISK-23692
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The iax2_best_codec function was changed to convert the formats
into a format compatibilities structure and grab the first
format from it. The resulting order differs from the previous
order of iax2_best_codec which causes unexpected formats to
get chosen (such as g723).
This commit brings back the old behavior of iax2_best_codec by
having a specified preference list.
Review: https://reviewboard.asterisk.org/r/3835/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After merging the media formats branch, chan_iax2 was discarding
codec preferences for the purpose of choosing which codec a
channel would use once a call started. This patch restores the
Asterisk 1.8-12 codec choice behaviors.
ASTERISK-23958 #close
Review: https://reviewboard.asterisk.org/r/3800/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
progressinband=never in sip.conf is easily defeated if an onward trunk sends a
progress indication of its own. This is almost certain to happen if the onward
trunk is ISDN or IAX as these technologies send a progress indication even if
early media is not required. This progress message is passed to the caller,
and causes the "never" option to be rather badly named.
This patch changes the behaviour of this setting in the following ways:
1) In sip_write(), do not pass the media unless we have either progressed
beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early
media is both set-up and wanted. This helps resolve double-ringing on some
buggy handsets.
2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but
SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly
enabling early media. Avoid sending double ring indications.
NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch
to also encapsulate the fact that a channel has *sent or received* a 183
Progress indication. This makes the updated code in sip_write() much more
simple.
Review: https://reviewboard.asterisk.org/r/3700
ASTERISK-23972 #close
Reported by: Steve Davies
patches:
inband_never_present_early_media2 uploaded by Steve Davies (License 5012)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most channel drivers let you specify a default accountcode to be set on
channels associated with a particular peer/endpoint/object. Prior to this
patch, chan_pjsip/res_pjsip did not support such a setting.
This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'.
When a channel is created that is associated with an endpoint with this value
set, the channel will automatically have its accountcode property set to the
value configured for the endpoint.
Review: https://reviewboard.asterisk.org/r/3724/
ASTERISK-24000 #close
Reported by: Matt Jordan
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Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Walter Doekes
Matthew Jordan
Corey Farrell
Jonathan Rose
Joshua Colp
Mark Michelson
Scott Griepentrog
Michael L. Young
Paul Belanger
Kinsey Moore
Damien Wedhorn