We were still getting crashes after the first fix. Somehow we receive a
non-2xx final response before we get a 200 final response. With the
failure response we had already cleaned up and destroyed some data
structures. When the unexpected 200 response comes in we crash.
* Add protection code to prevent processing another final T.38 reINVITE
response.
ASTERISK-27944
Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74
The allow_unqualified_fetch option for the sorcery realtime backend
blocked actually fetching all rows when the option is set to warn.
* Made issue a warning and actually do the request when
allow_unqualified_fetch=warn is set.
Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312
Compiling without SRTP support installed resulted in some unused variable
warnings. These warnings also showed that the srtp variable was obtained
and passed around some functions but not really used even when a system
has SRTP installed.
Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc
If in the initial sdp the caller doesn't include the line
a=rtcp-mux
Then asterisk shoud not include rtcp-mux in the response regardless
of rtcp-mux being enabled on the endpoint
ASTERISK-28007 #close
Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7
sip_options_get_endpoint_state_compositor_state leaked a reference to
the first available endpoint state compositor that was found.
Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7
This change adds the "party" parameter to the Remote-Party-ID header
which indicates which party the header information is applicable
to. In Asterisk this is determined on whether we are the calling
or called party. This is added to improve interoperability with some
implementations.
ASTERISK-28006
Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca
When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.
Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.
ASTERISK-27398
Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e
The authors of PJProject undef s_addr because of some issue in Microsoft
Windows. However in Oracle Solaris, s_addr is not a structure member, but
defined to map to the real structure member.
Updates the patch from ASTERISK_20366
ASTERISK-27997
Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11
* Don't include pjlib.h twice in res_pjsip.h
* Consistently use #include <> form for pjproject includes.
(pjsip.h and pjlib.h)
Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7
If asterisk offer an endpoint with SRTP and that endpoint respond
with non srtp, in that case channel(rtp,secure,audio) reply wrong
status.
Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key:
Currently this flag has being set redundantly. In either case identical
or different remote_key this flag has being set. So if we
don't set it while we receive identical remote_key or non SRTP SDP
response then we can take decision of srtp use by using that flag.
ASTERISK-27999
Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7
This change removes a sorcery lookup for retrieving all
contacts at the end of the registration process by keeping
track of the contacts that are added/updated/deleted.
This ensures at the end of the process the container of
contacts we have is the current state.
Pool usage has also been reduced by allocating one for
usage throughout the handling of a REGISTER and resetting
it to a clean state. This ensures that in most cases
we allocate once and just reuse it.
ASTERISK-28001
Change-Id: I1a78b2d46f9a2045dbbff1a3fd6dba84b612b3cb
This patch adds regular expression support to make the identify section's
match_header option more useful when attempting to match complex headers
like the 'To' or 'From' headers. The 'From' header has variable
components such as the tag parameter that you cannot predict. To specify
a regular expression put slashes around the regular expression in place of
the header value.
[identify-alice]
type=identify
endpoint=alice
match_header=From: /<sip:alice@127\\.0\\.0\\.1>/
* Added regex support to match_header so you could match a 'To' header
among other complex headers.
Fixed reported crashes when trying to match special headers like 'Contact'.
The identify section's match_header method used code that assumed you were
matching a generic header. Any other type of header could cause a crash
if the header structure variant did not match the generic header enough.
* Made use code that will work for any header type instead of code
specific to generic headers.
Other fixes while in the area:
* Made check all headers of the requested name.
* Added some more sanity checks to the configured identify matching
options when applying the configuration.
ASTERISK-27548
Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1
The alembic for the PJSIP subscription persistence table has the
"prune_on_boot" field as a boolean. While in Asterisk we are
tolerant of many different definitions of true and false in the
database we only accept "yes" and "no". This change makes the
field treated as a yes/no instead of an integer, thus storing
"yes" and "no" instead of "1" and "0".
Change-Id: Ic8b9211b36babefe78f70def6828a135a6ae7ab6
OpenSSL is an optional external library and should stay optional even when
Developer Mode is configured.
ASTERISK-27990
Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b
When realtime text packets are to be sent, the text is accumulated in a
buffer and sent regularly by a timer. It can happen that commands such as
a backspace, CR, or LF get merged with regular text. This breaks some
UAs.
The proposed change:
* We test if the current packet contains a command. If so we send the
buffer immediately.
* We test if the buffer contained a command. If so we send the buffer
immediately.
* We accumulate the text (or the command) in the buffer.
ASTERISK-27970
Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462
If a SIP MESSAGE is triggered for an endpoint that is currently not registered
- and therefore has no valid contact associated - an error message was logged.
Since this is a valid request in a valid use cases this is now changed to a
warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list.
Change-Id: I55eb62d2712818a58c7532119dec288bd98cf0c0
A change recently went in which disabled the built-in PJSIP
keepalive. This defaulted to 90 seconds and kept TCP/TLS
connections alive. Disabling this functionality has resulted
in a behavior change of not doing keepalives by default resulting
in TCP/TLS connections dropping for some people.
This change makes our default keepalive interval 90 seconds
to match the previous behavior and preserve it.
ASTERISK-27978
Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6
* Use the replacement function ast_sip_push_task_wait_servant() instead of
the deprecated ast_sip_push_task_synchronous().
Change-Id: I145b550ba7054640c7faa3b644e63137f505c612
Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.
If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.
According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.
Also added additional functionality to ast_data_buffer, along with some
testing.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
ASTERISK-27810 #close
Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d
A problem I've seen countless times is a global or system section
for PJSIP not getting applied. This is inevitably the result of
the "type=" line missing. This change alleviates that problem.
The ability to specify an explicit section name has been
added to res_sorcery_config. If the configured section
name matches this and there are no unknown things configured
the section is taken as being for the given type.
Both the PJSIP "global" and "system" types now support this
so you can just name your section "global" or "system" and it
will be matched and used, even without a "type=" line.
ASTERISK-27972
Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893
I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.
ASTERISK-27965
Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7
SRTP SDES key lifetime support was added in ASTERISK_17899.
In that addition, the minimum key lifetime to be accepted was
set at the 10 hours @ 20ms/packet = 1800000 packets.
The firmware in the obi1xx ATA uses a hardcoded lifetime of
2^20 packets.
Lower the limit to 2^20 to support a wider field of clients.
ASTERISK-27967 #close
Change-Id: I81a0703c595a0c9101dfdf02300149a3cc39bf94
Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.
Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label"
attribute to the current sdp.
Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5
Keep track if ICE candidates were in the SDP offer & only put them
in the corresponding SDP answer if the offer condaind ICE candidates
ASTERISK-27957 #close
Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92
This commit adds a new function to res_parking.
This function, PARK_GET_CHANNEL allows the retrieval
of the channel name of the channel occupying the parking slot.
ASTERISK-22825 #close
Change-Id: Idba6ae55b8a53f734238cb3d995cedb95c0e7b74
