Commit Graph

28327 Commits

Author SHA1 Message Date
Joshua Colp
b83c9ff2ad Merge "pjproject_bundled: Improve reliability of pjproject download" into 13 2016-11-21 06:23:58 -06:00
Joshua Colp
5c9680da9d Merge "main/app.c: Transmit Silence on ControlPlayback pause" into 13 2016-11-21 04:46:42 -06:00
Joshua Colp
9e5f76c1e1 Merge "build: Various OpenBSD issues" into 13 2016-11-18 12:37:59 -06:00
Joshua Colp
51d2d7c359 Merge "Bump ARI version to 1.10.0" into 13 2016-11-18 12:35:45 -06:00
Mark Michelson
cb624b10ae Bump ARI version to 1.10.0
The video-related bridge changes mean that the version needs to be
bumped.

Change-Id: I41c4495068562bef03aa76728f188b8ac4bd393d
2016-11-18 10:56:50 -05:00
George Joseph
2a40c3a867 pjproject_bundled: Improve reliability of pjproject download
The download process now has a timeout which will cause wget to retry
if it stops retrieving data for 5 seconds and fetch and curl to timeout
if the whole retrieval take smore than 30 seconds.

If the tarball retrieval works, the MD5SUM file is retrieved from
the downloads site and the md5 checksum is verified.

If either the tarball retrieval or MD5SUM retrieval fails, or the
checksums don't match, the entire process is retried once.  If it
fails again, any incomplete tarball is deleted.

.DELETE_ON_ERROR: was also added to the Makefile.  Not only does
this delete the tarball on failure, it till also delete corrupted
library files from the pjproject source directory should they
fail to build correctly.

Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and
Ubuntu 14.

Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1
2016-11-18 08:01:21 -05:00
misha
12c4e664bc main/app.c: Transmit Silence on ControlPlayback pause
ASTERISK-26562 #close

Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8
2016-11-17 12:28:25 -05:00
Mark Michelson
bde3d022a3 manager: update minor version
Based on bridge video AMI event changes, bump the minor version of AMI.

Change-Id: I02586bd6cafc0baa33ea98c2f75356c0f5e03435
2016-11-17 10:50:58 -06:00
zuul
3135a745e3 Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak." into 13 2016-11-16 23:20:10 -06:00
George Joseph
d3f921ed51 Merge "res_format_attr_opus: Fix fmtp generation." into 13 2016-11-16 22:41:00 -06:00
George Joseph
b213045fe4 build: Various OpenBSD issues
OpenBSD's 'find' doesn't take the -delete argument so you have to pipe
through 'xargs rm -rf'.

'echo -e' doesn't like \t starting a line. It just prints 't' which
causes the libasteriskpj.exports file to be garbage.  They were just
cosmetic so they were removed.

librt doesn't exist so the link of libasteriskpj.so fails. It's not
actually needed for linux anyway so -lrt was removed from the link.

res_rtp_asterisk was failing to load because of an undefined
DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if
so DTLSv1_method is used instead.

ASTERISK-26608

Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c
2016-11-16 19:34:20 -07:00
George Joseph
c877801e5d Merge "channel: Fix issues in hangup scenarios caused by frame deferral" into 13 2016-11-16 17:42:17 -06:00
George Joseph
404596b790 channel: Fix issues in hangup scenarios caused by frame deferral
ASTERISK-26343

Change-Id: I06dbf7366e26028251964143454a77d017bb61c8
2016-11-16 16:41:42 -07:00
Joshua Colp
19ca0e7fc8 Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded." into 13 2016-11-16 17:40:36 -06:00
zuul
b745c326c2 Merge "res/ari/resource_bridges: Add the ability to manipulate the video source" into 13 2016-11-16 16:48:14 -06:00
Mark Michelson
2c031b67d3 res_format_attr_opus: Fix fmtp generation.
res_format_attr_opus assumed that the string being passed into it was
empty. It tried to determine if the only thing it had written was

a=fmtp:<num>

And if it had, it would reset the string. Its calculation was off when
working with chan_sip, though. chan_sip passes the entire built SDP
rather than an empty string. This resulted in always putting an empty
fmtp line in the SDP.

ASTERISK-26520 #close
Reported by scgm11

Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5
2016-11-16 15:42:39 -06:00
Joshua Colp
4c1b9fbefe Merge "Revert "Revert "channel: Use frame deferral API for safe sleep.""" into 13 2016-11-16 15:39:00 -06:00
Joshua Colp
d0a72d3779 Merge "Revert "Revert "autoservice: Use frame deferral API""" into 13 2016-11-16 15:38:55 -06:00
zuul
36b59ee9c1 Merge "Revert "Revert "AGI: Only defer frames when in an interception routine.""" into 13 2016-11-16 15:06:25 -06:00
zuul
f68790d46a Merge "Revert "Revert "Add API for channel frame deferral.""" into 13 2016-11-16 15:06:24 -06:00
zuul
f4f152f35c Merge "apps/app_echo: Only relay a single video source change frame" into 13 2016-11-16 15:06:23 -06:00
Richard Mudgett
ed0f1afc8c codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.
When Opus is negotiated but not loaded, the log is spammed with messages
because the system does not know how to calculate the number of samples in
a frame.

* Suppress the warning by supplying a function that assumes 20ms of
samples in the frame.  For pass through support it doesn't really seem to
matter what number of samples is returned anyway.

ASTERISK-26605 #close

Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f
2016-11-16 13:49:23 -06:00
Joshua Colp
4c3d25875f Merge "Add X.509 subject alternative name support to TLS certificate verification." into 13 2016-11-16 13:14:42 -06:00
Joshua Colp
6672fd5faa Merge "cli: Fix ast_el_read_char to work with libedit >= 3.1" into 13 2016-11-16 12:50:15 -06:00
Richard Mudgett
e632222bc4 res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.
Responding to authentication challenges leaks PJSIP memory pools.

The leak was introduced with a pjproject 2.5.5 API change.
https://trac.pjsip.org/repos/ticket/1929 changed the API usage of
pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to
clean up cached authentication allocations that get allocated with
pjsip_auth_clt_reinit_req().

ASTERISK-26516 #close

Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8
2016-11-16 12:02:10 -06:00
Joshua Colp
c4baa50f9a Merge "file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type" into 13 2016-11-16 11:12:16 -06:00
George Joseph
c92dcc76da file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type
One of the code paths in __ast_file_read_dirs will only get executed if
the OS doesn't support dirent->d_type OR if the filesystem the
particular file is on doesn't support it.  So, while standard Linux
systems support the field, some filesystems like XFS do not.  In this
case, we need to call stat() to determine whether the directory entry
is a file or directory so we append the filename to the supplied
directory path and call stat.  We forgot to truncate path back to just
the directory afterwards though so we were passing a complete file name
to the callback in the dir_name parameter instead of just the directory
name.

The logic has been re-written to only create a full_path if we need to
call stat() or if we need to descend into another directory.

Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba
2016-11-15 19:20:16 -07:00
Maciej Szmigiero
7b96e8cc3d Add X.509 subject alternative name support to TLS certificate
verification.

This way one X.509 certificate can be used for hosts that
can be reached under multiple DNS names or for multiple hosts.

Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>

ASTERISK-25063 #close

Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f
2016-11-15 20:56:43 +01:00
Matt Jordan
0790aa528a pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS
The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how
many pairs of local/remote candidates will be made. If for some reason
we reach this upper bound, ICE will generally fail and no media will
flow between the browser and Asterisk.

This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of
pairs of candidates we'd theoretically allow, which is
PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied
PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame
Docker), this is far too low to allow WebRTC calls to succeed.

Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed
even when the system Asterisk was running on had quite a few virtual
interfaces.

Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55
2016-11-14 17:02:17 -05:00
Matt Jordan
993a6f96c7 apps/app_echo: Only relay a single video source change frame
In 9785e8d0, app_echo was updated to relay video source updates to the
channel for the purposes of displaying video in WebRTC tests.
Unfortunately, this can cause a Kafkaesque nightmare if two or more
Local channels are in a bridge together where their ends are in
app_echo. When this situation occurs, a video update sent into app_echo
will cause the video update to be relayed to the other Local channels,
causing another round of video updates, etc. In not much time at all,
the channel length queues will be overwhelmed, channel alert pipes will
fail, and all hell will break loose as Asterisk merrily continues to
throw more video update requests onto the channels.

This patch updates app_echo to *only* relay a single video update. Once
a video update has been made, all further video updates are dropped.
This meets the intended purpose of the original patch: if we get a video
update and we're in app_echo, go ahead and ask the sender to update
themselves. However, once we've got that video stream sync'd up, don't
keep spamming the world.

Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74
2016-11-14 17:02:09 -05:00
Matt Jordan
d23b4af477 res/ari/resource_bridges: Add the ability to manipulate the video source
In multi-party bridges, Asterisk currently supports two video modes:
 * Follow the talker, in which the speaker with the most energy is shown
   to all participants but the speaker, and the speaker sees the
   previous video source
 * Explicitly set video sources, in which all participants see a locked
   video source

Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.

This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
    Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
    Removes any explicit video source, and sets the video mode to talk
    detection

ASTERISK-26595 #close

Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
2016-11-14 17:02:00 -05:00
George Joseph
404a62eeee Revert "Revert "channel: Use frame deferral API for safe sleep.""
This reverts commit 58c88cfbaa.

Change-Id: I72692e2b2e83ef6da9390075ff20b138b2c374b6
2016-11-14 14:22:43 -06:00
George Joseph
09d8febc91 Revert "Revert "autoservice: Use frame deferral API""
This reverts commit 1df434e2b4.

Change-Id: Id2b8a8bccbb4bbdd82b792275d4cd6f32563e401
2016-11-14 14:22:21 -06:00
George Joseph
ffad2b44df Revert "Revert "AGI: Only defer frames when in an interception routine.""
This reverts commit 6be5d8de0d.

Change-Id: I4b548137f52ae0686d8f09e21496b778d1c6a797
2016-11-14 14:21:58 -06:00
George Joseph
2fefb6187f Revert "Revert "Add API for channel frame deferral.""
This reverts commit 6b5a7ced13.

Change-Id: I61d1dbb2e69e1977f684b7dfc8e98211024e1cd1
2016-11-14 15:21:26 -05:00
zuul
a58d359701 Merge "res_pjsip.c: Rework endpt_send_request() req_wrapper code." into 13 2016-11-14 12:44:41 -06:00
George Joseph
5e0c224043 cli: Fix ast_el_read_char to work with libedit >= 3.1
Libedit 3.1 is not build with unicode on as a default and so the
prototype for the el_gets callback changed from expecting a char buffer
to accepting a wchar buffer.  If ast_el_read_char isn't changed,
the cli reads garbage from teh terminal.

Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and
updated ast_el_read_char to use the HAVE_ define to detemrine whether
to use char or wchar.

ASTERISK-26592 #close

Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a
2016-11-14 13:20:26 -05:00
Joshua Colp
ec17e96784 Merge "Fix closing rtp ports after call finished in chan_unistim." into 13 2016-11-14 08:05:38 -06:00
Joshua Colp
d5774005fe Merge "res_pjsip: Fix tdata leaks in off nominal paths." into 13 2016-11-14 06:15:44 -06:00
Igor Goncharovskiy
3faca1d4ff Fix closing rtp ports after call finished in chan_unistim.
Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.

Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc
2016-11-11 23:02:11 -05:00
Joshua Colp
10e64a5fb8 Merge "res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp." into 13 2016-11-11 15:17:55 -06:00
zuul
cf08676c00 Merge "build: Fix default values for some SANITIZER options" into 13 2016-11-10 22:09:02 -06:00
Richard Mudgett
412d43fa21 res_pjsip.c: Rework endpt_send_request() req_wrapper code.
* Don't hold the req_wrapper lock too long in endpt_send_request().  We
could block the PJSIP monitor thread if the timeout timer expires.
sip_get_tpselector_from_endpoint() does a sorcery access that could take
awhile accessing a database.  pjsip_endpt_send_request() might take awhile
if selecting a transport.

* Shorten the time that the req_wrapper lock is held in the callback
functions.

* Simplify endpt_send_request() req_wrapper->timeout code.

* Removed some redundant req_wrapper->timeout_timer->id assignments.

Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9
2016-11-10 16:17:33 -06:00
Richard Mudgett
2e7fc56d3c res_pjsip: Fix tdata leaks in off nominal paths.
Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b
2016-11-10 16:14:55 -06:00
Richard Mudgett
da68b185b3 res_pjsip_registrar_expire.c: Remove extra linefeed in debug message.
Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94
2016-11-10 14:23:46 -06:00
Joshua Colp
b70eb07c53 res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp.
When optimistic SRTP was on it was possible for us to still
set up a call without an audio stream if an offer was received
with required SRTP.

This change makes it so this scenario will now fail with a 488
response.

ASTERISK-26575

Change-Id: I7d14187037681f48879bd20319ac79d0877318f3
2016-11-10 16:57:49 +00:00
Joshua Colp
71dc333565 app_queue: Add mention of 'ABANDON' variable to CHANGES.
ASTERISK-26558

Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e
2016-11-10 14:33:41 +00:00
George Joseph
ccc8a37584 Merge "Revert "autoservice: Use frame deferral API"" into 13 2016-11-10 07:42:37 -06:00
George Joseph
07e7ac5119 Merge "Revert "Add API for channel frame deferral."" into 13 2016-11-10 07:42:36 -06:00
George Joseph
06045fc29e Merge "Revert "AGI: Only defer frames when in an interception routine."" into 13 2016-11-10 07:42:36 -06:00