We should not do flood detection on video RTP streams. Video RTP streams
are very bursty by nature. They send out a burst of packets to update the
video frame then wait for the next video frame update. Really only audio
streams can be checked for flooding. The others are either bursty or
don't have a set rate.
* Added code to selectively disable packet flood detection for video RTP
streams.
ASTERISK-27440
Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
A couple of places were setting the status to "UNKNOWN" when qualifies were
being disabled. Instead this should be set to the "CREATED" status that
represents when a contact is given (uri available), but the qualify frequency
is set to zero so we don't know the status.
This patch updates the relevant places with "CREATED". It also updates the
"CREATED" status description (value shown in CLI/AMI/ARI output) to a value
of "NonQualified"/"NonQual" as this description is hopefully less confusing.
ASTERISK-27467
Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89
Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.
This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.
Related to ASTERISK~26806, but does not completely resolve it.
Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
Previous commits maintained compatibility with older remote console
clients as well as maintaining all API's.
Remove the following compatibility code:
* ast_cli_generatornummatches.
* Remote command "_command nummatches".
* Sorting / duplicate removal by remote console.
Change-Id: I59e6ce94fa57ae564888442049695f7e46746437
Some completion generators are very inefficent due to the way CLI
requests matches one at a time. ast_cli_completion_add can be called
multiple times during one invokation of a CLI generator to add all
results without having to reinitialize the search state for each match.
Change-Id: I73d26d270bbbe1e3e6390799cfc1b639e39cceec
The ability to add to localized storage cannot be supported by
ast_cli_generator. The only calls to ast_cli_generator should be by
functions that need to proxy the CLI generator, for example 'cli check
permissions' or 'core show help'.
* ast_cli_generatornummatches now retrieves the vector of matches and
reports the number of elements (not including 'best' match).
* test_substitution retrieves and iterates the vector.
Change-Id: I8cd6b93905363cf7a33a2d2b0e2a8f8446d9f248
This adds the printf attribute and changes 'fmt' from 'const void *' to
'const char *'. This resolves a warning from some compiler for
vsnprintf needing a literal string for format.
Change-Id: I71c33a8262590042ee451e1146760c10bb22fb78
Previously for PJSIP the local address of WebSocket connections
was set to the remote address. For logging purposes this is
not particularly useful.
The WebSocket API has been extended to allow the local
address to be queried and this is used in PJSIP to set the
local address to the correct value.
The PJSIP HEP support has also been tweaked so that reliable
transports always use the local address on the transport
and do not try to (wrongly) guess. As they are connection
based it is impossible for the source to be anything else.
ASTERISK-26758
ASTERISK-27363
Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca
Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
This is a rewrite of ast_cli_completion_matches using a vector to build
the list. The original function calls the vector version, NULL
terminates the vector and extracts the elements array.
One change in behavior the results are now sorted and deduplicated. This
will solve bugs where some duplicate checking was done before the list
was sorted.
Change-Id: Iede20c5b4d965fa5ec71fda136ce9425eeb69519
* AST_VECTOR_STEAL_ELEMENTS - steal the array of elements for use
with non-vector code.
* struct ast_vector_string - a vector of 'char *'.
Change-Id: I104d1b204be03fccf67e02a195596adcb5ab1e42
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Sample asterisk.conf setting:
[options]
cache_media_frames=no
ASTERISK-27413
Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00
This change causes the configure script to fail if the C compiler does
not support both function attributes constructor and destructor. These
were already required as modules cannot function without these attributes
and Asterisk requires modules.
This also has AST_GCC_ATTRIBUTE set a variable
ax_cv_have_func_attribute_$1. This is the same variable name used by
autoconf-archive's AX_GCC_FUNC_ATTRIBUTE, used for the same purpose.
Change-Id: Id68e8a1447f2a6d707c54b56350e7bfdb33fb663
This adds FD tracking for the following functions:
* eventfd
* timerfd_create
* socketpair
* accept
ASTERISK-27404
Change-Id: Id6848fe904ade2d34eb39d2a20bd6b223e1111fc
When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it. The most likely way it can return failure is for an
allocation failure. If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.
* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure. That is much more useful than either an uninitialized pointer
or a pointer that has already been freed. Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.
* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().
* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.
Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23
This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.
Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.
ASTERISK-27395
Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
This check is being added to make it easier for end-users of third party
open source Opus modules. This was removed by ASTERISK-26426 but only
the module needed to be removed.
Change-Id: I62b9cd0c4fa8a77596ab0e042948a643a1152677
Replace 'needsreload' argument with a 'type' argument to specify which
type of modules you want completion. This provides more accurate CLI
completion for load and unload commands.
* 'module unload' now excludes modules that have active references or are
not running.
* 'module load' now excludes modules that are already running.
* 'core set debug [atleast] <level> [module]' shows running modules only.
ASTERISK-27378
Change-Id: Iea3e00054461484196c46f688f02635cc886bad1
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.
This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.
ASTERISK-27379
Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.
ASTERISK-27206
Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd
The configure option to disable XML documentation does not currently
work. This patch makes it effective, but also causes an ABI change by
removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs
from being automatically generated, but they can still be manually
generated with 'make doc/core-en_US.xml'.
ASTERISK-26639
Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44
This makes the 'bt' parameter unconditional for ast_store_lock_info and
ast_remove_lock_info. The 'bt' parameter is unused when HAVE_BKTR is
undefined.
Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453
Beanstalkd is a simple to use job queue. It provides a means to
create multiple job queues called "tubes". Each tube can store
multiple jobs, with varying priorities with the queue. Queue
processing is available via a simple TCP socket or via well defined
libraries, avaialble at
https://github.com/kr/beanstalkd/wiki/client-libraries
This module is based upon the beanstalk-client library, available
for download at: https://github.com/deepfryed/beanstalk-client
Change-Id: I5fe4089a34ab3b39230786d9bbfddafa56715f48
Add bridge_features structure to bridge creation. Specifically, this
implements mute and DTMF suppression, but others should be able to be
easily added to the same structure.
ASTERISK-27322 #close
Reported by: Darren Sessions
Sponsored by: AVOXI
Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61
* Store weak proxy objects in instances container.
* Remove special unreference function and replace with macro that calls
ao2_cleanup.
* Add REF_DEBUG information to ast_sorcery_open.
Change-Id: I5a150a4e13cee319d46b5a4654f95a4623a978f8
This function finds a weak proxy in an ao2_container and returns the
real object associated with it.
Change-Id: I9da822049747275f5961b5c0a7f14e87157d65d8
Copy the list of weakproxy callbacks to temporary memory so they can be
run without holding the weakproxy lock.
Change-Id: Ib167622a8a0f873fd73938f7611b2a5914308047
When "rewrite_contact" is enabled, the "max_contacts" count option can
block re-registrations because the source port from the endpoint can be
random. When the re-registration is blocked, the endpoint may give up
re-registering and require manual intervention.
* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire. The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one. The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.
ASTERISK-27192
Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b
Use temporary variable to prevent multiple evaluations of elem argument.
This resolves a memory leak in res_pjproject startup.
ASTERISK-27317 #close
Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d
This provides better information to REF_DEBUG log for troubleshooting
when the system is unable to unload res_pjsip.so during shutdown due to
module references.
ASTERISK-27306
Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612
res_pjsip and res_pjsip_session had circular references, preventing both
modules from shutting down.
* Move session supplement registration to res_pjsip.
* Use create internal functions for use by pjsip_message_filter.c.
ASTERISK-27306
Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b
Somewhere along the way we lost the ability to debug individual
source files. For modules, this wasn't a big deal but all the
source files in ./main are in the one "core" module so debugging
individual core capabilities was almost impossible.
* Added a test to DEBUG_ATLEAST that also checks __FILE__ instead
of just module name. Any source file will work even if it's in
a module subdirectory.
Change-Id: Icc0af41837f3b1679dec7af21fa32cd1f7469f6e
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
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
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
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
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
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
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
* Fix deadlock in
bridge_softmix.c:softmix_bridge_stream_topology_changed() between
bridge_channel and channel locks.
* The new bridge technology topology change callbacks must be called with
the bridge locked. The callback references the bridge channel list, the
bridge technology could change, and the bridge stream mapping is updated.
ASTERISK-27212
Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be
* ast_channel_request_stream_topology_change() must not be called with any
channel locks held.
* ast_channel_stream_topology_changed() must be called with only the
passed channel lock held.
ASTERISK-27212
Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691
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
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
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
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
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
AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0
Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago
Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7
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 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
This adds support for parsing timelen values from config files. This
includes support for all flags which apply to PARSE_INT32. Support for
this parser is added to ACO via the OPT_TIMELEN_T option type.
Fixes an issue where extra characters provided to ast_app_parse_timelen
were ignored, they now cause an error.
Testing is included.
ASTERISK-27117 #close
Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554
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
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
This change fixes a few things uncovered during SFU testing.
1. Unreal channels incorrectly forwarded video frames when
no video stream was present on them. This caused a crash when
they were read as the core requires a stream to exist for the
underlying media type. The Unreal channel will now ensure a
stream exists for the media type before forwarding the frame
and if no stream exists then the frame is dropped.
2. Mapping of frames during bridging from the stream number of
the underlying channel to the stream number of the bridge was
done in the wrong location. This resulted in the frame getting
dropped. This mapping now occurs on reading of the frame from
the channel.
3. Bridging was using the wrong ast_read function resulting in
it living in a non-multistream world.
4. In bridge_softmix when adding new streams to existing channels
the wrong stream topology was copied resulting in no streams
being added.
Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8
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
Richard Mudgett
Kevin Harwell
Jenkins2
Sean Bright
Corey Farrell
George Joseph
Joshua Colp
Alexander Traud
Nir Simionovich
Thomas Sevestre
Seán C McCord
Jean Aunis
Walter Doekes
Torrey Searle
Matthew Fredrickson