When dealing with a lot of video streams on WebRTC
the resulting SDPs can grow to be quite large. This
effectively doubles the maximum size to allow more
streams to exist.
The res_http_websocket module has also been changed
to use a buffer on the session for reading in packets
to ensure that the stack space usage is not excessive.
Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01
If an ACL is misconfigured in the realtime database (for instance, the
"rule" is blank) and Asterisk attempts to read the ACL, Asterisk will
crash.
ASTERISK-28978 #close
Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610
I noticed this while looking at another issue and brought
it up with Teluu. It was possible for an uninitialized timer
to be cancelled, resulting in the invalid timer id of 0
being placed into the timer heap causing issues.
This change is a backport from the pjproject repository
preventing this from happening.
Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7
We read beyond the end of the buffer when copying the string out of the
buffer when we used ast_copy_string() because the original string was
not null terminated. Instead switch to ast_strndup() which does not
exhibit the same behavior.
ASTERISK-28975 #close
Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a
Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.
Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.
Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.
Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1
session_on_rx_response wasn't checking for a NULL dialog before
attempting to get the invite session from it.
Change-Id: Id13534375966cc2eb7f2b55717c9813c63c10065
If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.
(Actually, commit ed615afb7e already
partially fixed it for shared_lastcall.)
ASTERISK-28951
Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6
Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.
No functional changes were made with this commit.
Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c
This commit adds the endpoint options required to control
Advanced Codec Negotiation.
incoming_offer_codec_prefs
outgoing_offer_codec_prefs
incoming_answer_codec_prefs
outgoing_answer_codec_prefs
The documentation may need tweaking and some additional edits
added, especially for the "answer" prefs. That'll be handled
when things finalize.
This commit is safe to merge as it doens't alter any existing
functionality nor does it alter the previous codec negotiation
work which may now be obsolete.
Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11
Currently when the pjsip making an outgoing request, it keep adding the
rport parameter in a request message as a default.
This causes unexpected rport handle at the other end.
Added option for disable this behaviour in the pjsip.conf.
This is a system option, but working as a gloabl option.
ASTERISK-28959
Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc
* ast_frame_subclass2str() and ast_frame_type2str() now return
a pointer to the buffer that was passed in instead of void.
This makes it easier to use these functions inline in
printf-style debugging statements.
* Added many missing control frame entries in
ast_frame_subclass2str.
Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc
Tracing through synchronous tasks was a little troublesome because
the new thread's stack counter reset to 0. This change allows
a synchronous task to set its trace level to be the same as the
thread that pushed the task. For now, the task's level has to be
passed in the task's data structure but a future enhancement to the
taskprocessor subsystem could automatically set the trace level
of the servant to be that of the caller.
This doesn't really make sense for async tasks because you never
know when they're going to run anyway.
Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825
Do not return error if the client received ping frame
while looking for a string and just wait for another frame.
ASTERISK-28958 #close
Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922
When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs on refresh unless
they had a shared codec between the two.
This patch makes it so whatever is set using PJSIP_MEDIA_OFFER
is used when creating the SDP for a refresh no matter what.
ASTERISK-28878 #close
Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6
This patch allows a user of AMI to now specify the type of message
content contained within by setting the 'Content-Type' parameter.
Note, the AMI version has been bumped for this change.
ASTERISK-28945 #close
Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb
The Streams API becomes the home for the core ACN capabilities.
These include...
* Parsing and formatting of codec negotation preferences.
* Resolving pending streams and topologies with those configured
using configured preferences.
* Utility functions for creating string representations of
streams, topologies, and negotiation preferences.
For codec negotiation preferences:
* Added ast_stream_codec_prefs_parse() which takes a string
representation of codec negotiation preferences, which
may come from a pjsip endpoint for example, and populates
a ast_stream_codec_negotiation_prefs structure.
* Added ast_stream_codec_prefs_to_str() which does the reverse.
* Added many functions to parse individual parameter name
and value strings to their respectrive enum values, and the
reverse.
For streams:
* Added ast_stream_create_resolved() which takes a "live" stream
and resolves it with a configured stream and the negotiation
preferences to create a new stream.
* Added ast_stream_to_str() which create a string representation
of a stream suitable for debug or display purposes.
For topology:
* Added ast_stream_topology_create_resolved() which takes a "live"
topology and resolves it, stream by stream, with a configured
topology stream and the negotiation preferences to create a new
topology.
* Added ast_stream_topology_to_str() which create a string
representation of a topology suitable for debug or display
purposes.
* Renamed ast_format_caps_from_topology() to
ast_stream_topology_get_formats() to be more consistent with
the existing ast_stream_get_formats().
Additional changes:
* A new function ast_format_cap_append_names() appends the results
to the ast_str buffer instead of replacing buffer contents.
Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56
Created new SCOPE_ functions that don't depend on RAII_VAR. Besides
generating less code, the use of the explicit SCOPE_EXIT macros
capture the line number where the scope exited. The RAII_VAR
versions can't do that.
* SCOPE_ENTER(level, ...): Like SCOPE_TRACE but doesn't use
RAII_VAR and therefore needs needs one of...
* SCOPE_EXIT(...): Decrements the trace stack counter and optionally
prints a message.
* SCOPE_EXIT_EXPR(__expr, ...): Decrements the trace stack counter,
optionally prints a message, then executes the expression.
SCOPE_EXIT_EXPR(break, "My while got broken\n");
* SCOPE_EXIT_RTN(, ...): Decrements the trace stack counter,
optionally prints a message, then returns without a value.
SCOPE_EXIT_RTN("Bye\n");
* SCOPE_EXIT_RTN_VALUE(__return_value, ...): Decrements the trace
stack counter, optionally prints a message, then returns the value
specified.
SCOPE_EXIT_RTN_VALUE(rc, "Returning with RC: %d\n", rc);
Create an ast_str helper ast_str_tmp() that allocates a temporary
ast_str that can be passed to a function that needs it, then frees
it. This makes using the above macros easier. Example:
SCOPE_ENTER(1, Format Caps 1: %s Format Caps 2: %s\n",
ast_str_tmp(32, ast_format_cap_get_names(cap1, &STR_TMP),
ast_str_tmp(32, ast_format_cap_get_names(cap2, &STR_TMP));
The calls to ast_str_tmp create an ast_str of the specified initial
length which can be referenced as STR_TMP. It then calls the
expression, which must return a char *, ast_strdupa's it, frees
STR_TMP, then returns the ast_strdupa'd string. That string is
freed when the function returns.
Change-Id: I44059b20d55a889aa91440d2f8a590865998be51
The outbound proxy for an AOR was not being applied to
any statically configured Contacts. This resulted in the
OPTIONS requests being sent to the wrong target.
This change sets the outbound proxy on statically configured
contacts once the AOR configuration is done being
applied.
ASTERISK-28965
Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0
Given a scenario where a module has a dependency on both
an external library and a module if the external library was
available and the module was not an infinite loop would
occur. This happened due to the code changing the dependecy
status to no failure on each dependency checking loop
iteration, resulting in the code thinking that it had
gone from no failure to failure each time triggering another
dependency check.
This change makes it so that the old dependency status is
preserved throughout the dependency checking allowing it to
determine that after the first iteration the dependency
status does not transition from no failure to failure.
ASTERISK-28930
Change-Id: Iea06d45d9fd6d8bfd068882a0bb7e23a53ec3e84
chan_sip handle_response() function, for a 400 response to an INVITE,
calls handle_response_invite() and does not generate ACK.
handle_response_invite() does not recognize 400 response and has no
default response processing for unexpected responses, thus it does not
generate ACK either.
The ACK on response repetition comes from handle_response() mechanism
"We must re-send ACKs to re-transmitted final responses".
According to code history, 400 response specific processing was
introduced with commit
"channels/chan_sip: Add improved support for 4xx error codes"
This commit added support for :
- 400/414/493 in handle_response_subscribe() handle_response_register()
and handle_response().
- 414/493 only in handle_response_invite().
This fix adds 400 response support in handle_response_invite().
ASTERISK-28957
Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad
A patch made a reference to the PJSIP_SC_NULL enumeration value, which
was added to pjproject 2.8 and above thus making it so Asterisk would
fail to compile with prior versions of pjproject.
This patch removes the reference, and instead initializes the value
to '0'.
ASTERISK-28886 #close
Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7
1) Fix memory-leaks
Added code to release ast_events extracted from corosync and stasis messages
2) Clean stasis cache when a member of the corosync cluster leaves the group
Added code to remove from the stasis cache of the members remained on the
group all the messages with the EID of the left member.
If the device states of the left member remain in the stasis cache of other
members, they will not be updated anymore and high priority cached values,
like BUSY, will take precedence over current device states.
3) Stop corosync event propagation when node is not joined to the group
Updated dispatch_thread_handler code to detect when asterisk is not joined
to the corosync group and added some condition in publish_event_to_corosync
code to send corosync messages only when joined.
When a node is not joined its corosync daemon can't send messages:
the cpg_mcast_joined function append new messages to the FIFO buffer until
it's full and then it blocks indefinitely.
In this scenario if the stasis_message_cb callback, registered by
res_corosync to handle stasis messages, try to send a corosync messages,
the thread of the stasis thread-pool will be blocked until the node join
the corosync cluster.
ASTERISK-28888
Reported by: Università di Bologna - CESIA VoIP
Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2
When stream support was added to Asterisk the stream state
was used inconsistently, resulting in odd behavior. This
was then standardized to be the state of a stream from the
perspective of Asterisk.
This change updates the StreamEcho dialplan application
to use the correct state, send only, since we are only
sending to the endpoint and not expecting them to send us
multiple video streams.
ASTERISK-28954
Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56
The change to how setvar works for various channels performed in
ASTERISK~23756 missed some required change in the dahdi channel,
where the variables are actually set while reading configuration.
This change should fix the issue.
ASTERISK-28955
Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274
When a re-INVITE is received we create a new set of
streams that are then swapped in as the active streams.
We did not preserve the SDP label from the previous
streams, resulting in the label getting lost.
This change ensures that if an SDP label is present
on the previous stream then it is set on the new stream.
ASTERISK-28953
Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445
The AMI action and CLI command did not take into account the properties
of full backend caching. This resulted in an expired object remaining
removed until a full backend update occurred, instead of having the
object updated when needed.
This change makes it so that the AMI action and CLI command for object
expire will now fail instead of putting the cache into an undesired
state. If full backend caching is enabled then only operations
which act on the entire cache are available.
ASTERISK-28942
Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4
Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is
sent, the caller ID will be checked to see if there is a certificate
that corresponds to it. If so, that information will be retrieved and an
Identity header will be added to the SIP message. The format is:
header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken
Header, payload, and signature are all BASE64 encoded. The public key
URL is retrieved from the certificate. Currently the algorithm and ppt
are ES256 and shaken, respectively. This message is signed and can be
used for verification on the receiving end.
Two new configuration options have been added to the certificate object:
attestation and origid. The attestation is required and must be A, B, or
C. origid is the origination identifier.
A new utility function has been added as well that takes a string,
allocates space, BASE64 encodes it, then returns it, eliminating the
need to calculate the size yourself.
Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4
Because ring_entry() is not called, outgoing->chan is not touched here
either.
ASTERISK-28950
ASTERISK-28644
Change-Id: I564613715dfaf45af868251eb75a451f512af90f
Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.
This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.
In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.
ASTERISK-28952
Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3
This patch makes the usual necessary changes when upgrading to a new
version pjproject. For instance, version number bump, patches removed
from third-party, new *.md5 file added, etc..
This patch also includes a change to the Asterisk pjproject Makefile to
explicitly create the 'source/pjsip-apps/lib' directory. This directory
is no longer there by default so needs to be added so the Asterisk
malloc debug can be built.
This patch also includes some minor changes to Asterisk that were a result
of the upgrade. Specifically, there was a backward incompatibility change
made in 2.10 that modified the "expires header" variable field from a
signed to an unsigned value. This potentially effects comparison. Namely,
those check for a value less than zero. This patch modified a few locations
in the Asterisk code that may have been affected.
Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to
check a minimum version of pjproject at compile time.
ASTERISK-28899 #close
Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81
When requesting a Local channel the requested stream topology
or a converted stream topology will now be placed onto the
resulting channels.
Frames written in on streams will now also preserve the stream
identifier as they are queued on the opposite channel.
Finally when a stream topology change is requested it is
immediately accepted and reflected on both channels. Each
channel also receives a queued frame to indicate that the
topology has changed.
ASTERISK-28938
Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186
If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly.
Fixed it to parse the channelId, other_channel_id parameter correclty.
ASTERISK-28948
Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157
The "value" passed in when setting an RTP property determines
whether it should be enabled or disabled. The RTP send and
receive retrans props did not examine this to know if the
buffers should be enabled. They assumed they always should be.
This change makes it so that the "value" passed in is
respected.
ASTERISK-28939
Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc
There are three states that an old stream can be in to allow
becoming a source stream in a new stream:
1. Removed
2. Inactive
3. Sendonly
This change adds the two missing ones, inactive and sendonly,
so if a stream transitions from those to a state where they are
providing video to Asterisk we properly re-negotiate the other
participants.
ASTERISK-28944
Change-Id: Id8256b9b254b403411586284bbaedbf50452de01
When fax_gateway_framehook is called and a gateway hasn't already
been started, the framehook gets the t38 state for both the current
channel and the peer. That call trickles down to the channel
driver which determines the state. If either channel is hung up
(or in the process of being hung up), the channel driver's tech_pvt
is going to be NULL which, in the case of chan_pjsip, will cause a
segfault.
* Added a hangup check for both the channel and peer channel
before starting a fax gateway.
* Added a check for NULL tech_pvt to chan_pjsip_queryoption
so we don't attempt to reference a tech_pvt that's already
gone.
ASTERISK-28923
Reported by: Yury Kirsanov
Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c
When send_events is enabled for a user, we were leaking a reference
to the bridge channel in confbridge_manager.c:send_message(). This
also caused the bridge snapshot to not be destroyed.
Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97
This patch fixes a few compile warnings/errors that now occur when using gcc
10+.
Also, the Makefile.rules check to turn off partial inlining in gcc versions
greater or equal to 8.2.1 had a bug where it only it only checked against
versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures
any version above the specified version is correctly compared.
Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9
If parameters were passed in the body as JSON to the
create route they were not being parsed before checking
to ensure that required fields were set.
This change moves the parsing so it occurs before
checking.
ASTERISK-28940
Change-Id: I898b4c3c7ae1cde19a6840e59f498822701cf5cf
You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri,
without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S).
ASTERISK-28936
Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a
Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an
INVITE, the Identity header is retrieved, parsing the message to verify
the signature. If any of the parsing fails,
AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this
caller ID. If verification itself fails,
AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in
the payload does not line up with the SIP signaling,
AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps
pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the
verification process.
A new config option has been added to the general section for
stir_shaken.conf. "signature_timeout" is the amount of time a signature
will be considered valid. If an INVITE is received and the amount of
time between when it was received and when it was signed is greater than
signature_timeout, verification will fail.
Some changes were also made to signing and verification. There was an
error where the whole JSON string was being signed rather than the
header combined with the payload. This has been changed to sign the
correct thing. Verification has been changed to do this as well, and the
unit tests have been updated to reflect these changes.
A couple of utility functions have also been added. One decodes a BASE64
string and returns the decoded string, doing all the length calculations
for you. The other retrieves a string value from a header in a rdata
object.
Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913
If a frame is written to a channel in a bridge we
would normally queue this frame up and the channel
thread would then act upon it. If this frame had no
stream mapping on the channel it would then be
discarded.
This change adds a check before the queueing occurs
to determine if a mapping exists. If it does not
exist then the frame is not even queued at all. This
stops a frame duplication from happening and from
the channel thread having to wake up and deal with
it.
Change-Id: I17189b9b1dec45fc7e4490e8081d444a25a00bda
In a particular fax gateway scenario whereby it would
have to translate using the read translation path on a
channel the frame being translated would be consumed.
When the frame is in the write path it is not permitted
to free the frame as the caller expects it to continue
to exist.
This change makes it so that the frame is only consumed
on the read path where it is acceptable to free it.
ASTERISK-28900
Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d
