Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic
Edition after accepting the audio request but declining the video one.
Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c
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
chan_pjsip_indicate was missing a case for the recently added
AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an
error and causing the call to be hung up instead of just ignoring
it.
ASTERISK-27260
Reported by: Daniel Heckl
Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80
When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.
ASTERISK-27257
Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df
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
Provide a way to get the contents of the the Request URI from the initial SIP
INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}")
ASTERISK-27278
Reported by: David J. Pryke
Tested by: David J. Pryke
Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e
This change makes it so that the conference recorder channel
that is created only contains audio formats and an audio stream.
This is because the underlying application used by ConfBridge to
record, MixMonitor, only allows recording audio.
Having additional streams (and in particular a video stream) can
result in clients needlessly renegotiating to add a video stream
that will never receive video.
Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0
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 Websocket implementation will steal the underlying stream of
TCP/TLS sessions. This results in an error message being output
about a stream not being present when in reality this is actually
fine.
This change moves it to a debug message instead.
Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094
* 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
Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.
ASTERISK-21399 #close
Reported by: Tzafrir Cohen
Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3
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