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
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
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
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
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
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 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
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
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
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
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
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
When writing tx messages to pcap files, Asterisk is using the wrong
pointer resulting in lots of wasted space. This patch fixes it to use
the correct pointer.
ASTERISK-28932 #close
Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23
files
fwrite() does return the number of elements written and not the
number of bytes. However asterisk is currently comparing the return
value to the size of the written element what means that asterisk logs
five WARNING messages on every packet written to the pcap file.
This patch changes the code to check for the correct value, which will
always be 1.
ASTERISK-28921 #close
Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2
When replacing the user portion of the Contact URI the code
was using the ephemeral pool instead of the tdata pool. This
could cause the Contact user value to become invalid after a
period of time.
The code will now use the tdata pool which persists for the
lifetime of the message instead.
ASTERISK-28794
Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5
While asterisk is filtering out the x-ast-orig-host parameter from the
contact on response messages, it is not filtering it out from the
request URI and the to header on SIP requests (for example INVITE).
ASTERISK-28884 #close
Change-Id: Id032b33098a1befea9b243ca994184baecccc59e
When a configuration file in Asterisk is loaded
information about it is stored such that on a
reload it is not reloaded if nothing has changed.
This can be problematic when an error exists in
a configuration file in PJSIP since the error
will be output at start and not subsequently on
reload if the file is unchanged.
This change makes it so that if an error is
encountered when res_sorcery_config is loading
a configuration file a reload will always read
in the configuration file, allowing the error
to be seen easier.
Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed
The flags of a previous selection could have been set within the
object 'srtp', for example, when the previous selection returned
failure after setting just 'some' flags. Now, not to clutter the
code, all possible flags are cleared first, and then the selected
flags are set as before.
ASTERISK-28903
Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee
Added two unit tests, one for signing and another for verifying.
stir_shaken_sign checks to make sure that all the required parameters
are passed in and then signs the actual payload. If a signature is
produced and a payload returned as a result, the test passes.
stir_shaken_verify takes the signature from a signed payload to verify.
This unit test also verifies that all the required information is passed
in, and then attempts to verify the signature. If verification is
successful and a payload is returned, the test passes.
Change-Id: I9fa43380f861ccf710cd0f6b6c102a517c86ea13
The PJSIP packet logger now has the following CLI commands:
pjsip set logger pcap <filename>
When used this will create a pcap file containing the incoming
and outgoing SIP packets, in unencrypted form.
pjsip set logger verbose <on / off>
This allows you to toggle logging to verbose on and off.
pjsip set logger host <IP/subnet mask> add
This allows you to add an additional IP address or subnet
mask to logging, allowing you to log multiple instead of
just a single IP address or all traffic.
The normal "pjsip set logger host" CLI command has also been
expanded to allow subnet masks as well.
ASTERISK-28895
Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e
A warning was triggered that there may be a problem regarding file
extension (which is correct and should not be set anyway). The warning
also appeared if there was dot within the path itself.
E.g.
[sales-queue-hold]
mode=playlist
entry=/var/www/domain.tld/moh/funky_music
The music played correctly but you get a warning message.
Now there will be a check if the position of a potential dot character
is after the last position of a slash character. This dot charachter
will be treated as a extension naming. Dots within the path then ignored.
ASTERISK-28892
Reported-By: Nicholas John Koch
Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e
This change adds the same variable functionality that
is available for originating a channel to the create
call. Now when creating a channel you can specify
dialplan variables to set instead of having to do another
API call.
ASTERISK-28896
Change-Id: If13997ba818136d7c070585504fc4164378aa992
1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups
when an IPV6 transport type has been requested.
2. Rename all occurrences of pjsip_transport_get_type_name to
pjsip_transport_get_type_desc. This ensures that the log/debug info
shows whether the transport is IPv6 or IPv4.
3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport
types. This results in invalid values. Use a bitwise or instead.
ASTERISK-26780
Patches:
pjsip_resolver.c uploaded by Peter Sokolov (License #7070)
Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58
Adds the "STIR_SHAKEN" dialplan function and an API call to add a
STIR_SHAKEN verification result to a channel. This information will be
held in a datastore on the channel that can later be queried through the
"STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results
including identity, attestation, and verify_result. Here are some
examples:
STIR_SHAKEN(count)
STIR_SHAKEN(0, identity)
STIR_SHAKEN(1, attestation)
STIR_SHAKEN(2, verify_result)
Getting the count can be used to iterate through the results and pull
information by specifying the index and the field you want to retrieve.
Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82
Changed source and destination address fields in struct
pjsip_history_entry so that they are long enough to hold an IPv6
address.
ASTERISK-28854
Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e
There are a lot of moving parts in this patch, but the focus of it is on
the verification of the signature using a public key located at the
public key URL provided in the JSON payload. First, we check the
database to see if we have already downloaded the key. If so, check to
see if it has expired. If it has, redownload from the URL. If we don't
have an entry in the database, just go ahead and download the public
key. The expiration is tested each time we download the file. After
that, read the public key from the file and use it to verify the
signature. All sanity checking is done when the payload is first
received, so the verification is complete once this point is reached.
The XML has also been added since a new config option was added to
general (curl_timeout). The maximum amount of time to wait for a
download can be configured through this option, with a low value by
default.
Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c
Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined.
ASTERISK-28852 #close
Change-Id: I381718893b80599ab8635f2b594a10c1000d595e
Some places in Asterisk did not treat the formats on a stream
as immutable when they are.
The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.
An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.
ASTERISK-28846
Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe
Fixed it to copy the entire string from the requested endpoint body except tech-prefix.
ASTERISK-28847
Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25
This change fixes a few re-negotiation issues
uncovered with fax.
1. The fax support uses its own mechanism for
re-negotiation by conveying T.38 information in
its own frames. The new support for re-negotiating
when adding/removing/changing streams was also
being triggered for this causing multiple re-INVITEs.
The new support will no longer trigger when
transitioning between fax.
2. In off-nominal re-negotiation cases it was
possible for some state information to be left
over and used by the next re-negotiation. This
is now cleared.
ASTERISK-28811
ASTERISK-28839
Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2
MODULEINFO is checked while buidling/linking the module.
AST_MODULE_INFO is checked while loading/running the module.
ASTERISK-28838
Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042
When the receive buffer was flushed by a received packet while it
already contained a packet with the same sequence number, Asterisk
never left the while loop which tried to order the packets.
This change makes it so if the packet is in the receive buffer it
is retrieved and freed allowing the buffer to empty.
ASTERISK-28827
Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3
When the ast_data_buffer_put rejects to add a packet, for example because
the buffer already contains a packet with the same sequence number, the
payload will never be freed, resulting in a memory leak.
The data buffer will now return an error if this situation occurs
allowing the caller to free the payload. The res_rtp_asterisk module
has also been updated to do this.
ASTERISK-28826
Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1
By using pjproject to give us a list of candidates, and then filtering,
if the host has >32 addresses configured, then it is possible that we
end up filtering out all 32 of those, and ending up with no candidates
at all. Instead, get getifaddrs (which pjsip is using underlying
anyway) to retrieve all local addresses, and iterate those, adding the
first 32 addresses not excluded by the ICE ACL.
In our setup at any point in time We've got between 6 and 328 addresses
on any given system. The lower limit is the lower limit but the upper
limit is growing on a near daily basis currently.
Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
This reverts commit a3a2fbaec6.
Reason for revert: There is a lot of code that relies on the broken
behavior that this fixes.
Change-Id: I410c395a0168acbdaf89e616e3cb5e1312d190cb
When an AOR is modified endpoints are updated that reference
the AOR so they can start receiving updates and reflect the
correct state. If this is the case then we shouldn't change
the endpoint to be offline if it does not reference the AOR
but instead only when the endpoint is completely updated for
all its AORs.
ASTERISK-28056
patches:
pjsip_options-aor.diff submitted by jhord (license 6978)
Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164
This unit test runs through combinations of...
* Local codecs
* Remote Codecs
* Codec Preference
* Incoming/Outgoing
A few new APIs were created to make it easier to test
the functionality but didn't result in any actual
functional change.
ASTERISK_28777
Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164
Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.
* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)
* Add "call_direction" to res_pjsip_session.
* Update pjsip_session_caps.c to make the functions more generic
so they could be used for both incoming and outgoing.
* Update ast_sip_session_create_outgoing to create the
pending_media_state->topology with the results of
ast_sip_session_create_joint_call_stream().
* The endpoint "preferred_codec_only" option now automatically sets
AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.
* A helper function ast_stream_get_format_count() was added to
streams to return the current count of formats.
ASTERISK-28777
Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
This change provides functions that take in a JSON payload, verify that
the contents contain all the mandatory fields and required values (if
any), and signs the payload with the private key. Four fields are added
to the payload: x5u, attest, iat, and origid. As of now, these are just
placeholder values that will be set to actual values once the logic is
implemented for what to do when an actual payload is received, but the
functions to add these values have all been implemented and are ready to
use. Upon successful signing and the addition of those four values, a
ast_stir_shaken_payload is returned, containing other useful information
such as the algorithm and signature.
Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393
RFC5621 requires any content type with a Content-Disposition
with handling=required to be rejected with a 415 response
ASTERISK-28782 #close
Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4
When an outgoing channel is created a list of formats may
optionally be provided which is used as a request that the
formats be used if possible. If an endpoint is not configured
for any of the formats we ignore this request and use what is
configured. This has the side effect of also including other
stream types (such as video) that were not present in the
requested formats.
This change makes it so that the intention of the request is
preserved - that is if only an audio format is requested then
even if there is no joint audio format between the request and
the configuration we will still only place an audio stream in
the outgoing call.
ASTERISK-28787
Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f
This commit sets up some of the initial framework for the module and
adds a way to read the private key from the specified file, which will
then be appended to the certificate object. This works fine for now, but
eventually some other structure will likely need to be used to store all
this information. Similarly, the caller_id_number is specified on the
certificate config object, but in the end we will want that information
to be tied to the certificate itself and read it from there.
A method has been added that will retrieve the private key associated
with the caller_id_number passed in. Tab completion for certificates and
stores has also been added.
Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f
When examining a stream to determine hold/unhold information we
only care about the default audio stream. Other streams aren't
used for hold/unhold.
ASTERISK-28784
Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4
When the Asterisk receives 200 OK with invalid SDP,
the Asterisk/PJPROJECT terminating the session.
But if the channel was in the Bridge, Asterisk tries send
the Re-Invite before terminating the session.
And when the Asterisk sending the Re-Invite, it doesn't check
the SDP is NULL or not. This crashes the Asterisk.
Fixed it to close the session correctly if the UAS sends the
200 OK with wrong SDP.
ASTERISK-28743
Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da
The state of the default audio stream is used for hold/unhold so we
restrict it to sendrecv as the core does not handle when it changes as
a result of hold/unhold.
This restriction does not apply to other media types though so we now
only restrict it to audio. This allows the other default streams to
store their state at all values, and not just sendrecv and removed.
ASTERISK-28783
Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625
Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).
ASTERISK-28774
Reported-by: Michael Neuhauser
Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
A pure blacklist is not good enough, we need a whitelist mechanism as
well, and the simplest way to do that is to re-use existing ACL
infrastructure.
This makes it simpler to blacklist say an entire block (/24) except a
smaller block (eg, a /29 or even a /32). Normally you'd need to
recursively split the block, so if you want to blacklist a /24 except
for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I
feel that having an ACL instead of a blacklist only is clearer.
Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
bridge_p2p_rtp_write will forward rtp to the bridged rtp instance
without modifying the ssrc. However, it is not updating the SSRC
in the bridged rtp. Thus, when SSRC packets are generated, they
have the correct SSRC for the sender.
ASTERISK-28773 #close
Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478
If ICE support is enabled but not negotiated, the rtp->ice structure is
not being destroyed. This leads to Asterisk waiting for ICE to complete
instead of immediately starting the DTLS handshake, resulting in the
call leg having no RTP.
ASTERISK-28769 #close
Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27
This change introduce a CLI command for the RTP to display the general
configuration.
In the first step add the follow fields of the configurations:
- rtpstart
- rtpend
- dtmftimeout
- rtpchecksum
- strictrtp
- learning_min_sequential
- icesupport
Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f
Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.
This patch does the following:
Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.
Adds a new ast_sip_session_caps structure that's set for each session media
object.
Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.
ASTERISK-28756 #close
Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
The receive buffer will now grow if we end up flushing the
receive queue after not receiving the expected packet in time.
This is done in hopes that if this is encountered again the
extra buffer size will allow more time to pass and any missing
packets to be received.
The send buffer will now grow if we are asked for packets and
can't find them. This is done in hopes that the packets are
from the past and have simply been expired. If so then in
the future with the extra buffer space the packets should be
available.
Sequence number cycling has been handled so that the
correct sequence number is calculated and used in
various places, including for sorting packets and
for determining if a packet is old or not.
NACK sending is now more aggressive. If a substantial number
of missing sequence numbers are added a NACK will be sent
immediately. Afterwards once the receive buffer reaches 25%
a single NACK is sent. If the buffer continues to grow and
reaches 50% or greater a NACK will be sent for each received
future packet to aggressively ask the remote endpoint to
retransmit.
ASTERISK-28764
Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41
When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.
Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.
ASTERISK-28755 #close
Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f
Update the state of remote_hold immediately on receipt of remote
SDP so that the information is available when building the SDP
answer
ASTERISK-28754 #close
Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f
This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.
This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.
ASTERISK-28697
Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b
In order to retry outbound registrations for some situations, we
need access to the tdata from the original request. For instance,
for 401/407 responses we need it to properly construct the
subsequent request with the authentication. We also need it if
we're iterating over a DNS SRV response record set so we can skip
entries we've already tried.
We've been getting the tdata from the server response rdata and
transaction but that only works for the failures where there was
actually a response (4XX, 5XX, etc). For timeouts there's no
response and therefore no rdata or transaction from which to get
the tdata. When processing a single A/AAAA record for a server,
this wasn't an issue as we just retried that same server after the
retry timer expired. If we got an SRV record set for the server
though, without the state from the tdata, we just kept trying the
first entry in the set repeatedly instead of skipping to the next
one in the list.
* Added a "last_tdata" member to the client state structure to keep
track of the sent tdata.
* Updated registration_client_send() to save the tdata it used into
the client_state.
* Updated sip_outbound_registration_response_cb() to use the tdata
saved in client_state when we don't get a response from the
server. We still use the tdata from the transaction when we DO
get a response from the server so we can properly handle 4XX
responses where our new request depends on it.
General note on timeouts:
Although res_pjsip_outbound_registration skips to the next record
immediately when a timeout occurs during SRV set traversal, it's
pjproject that determines how long to wait before a timeout is
declared. As with other SIP message types, pjproject will continue
trying the same server at an interval specified by "timer_t1" until
"timer_b" expires. Both of those timers are set in the pjsip.conf
"system" section.
ASTERISK-28746
Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.
The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.
Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.
ASTERISK-28733
Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5
When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.
A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.
ASTERISK-28716
Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867
When moh_passthrough is used, asterisk is only generating invites
of type sendonly and sendrecv instead of taking fully into account
the on hold state of the local and remote parties
ASTERISK-28738 #close
Change-Id: Iaaad9fbc033cb14803d433b8a4071bc337047761
There was a race condition between client initiated DTLS setup, and handling
of server side ice completion that caused the underlying SSL object to get
cleared during DTLS initialization. If this happened Asterisk would be left
in a partial DTLS setup state. RTP packets were sent and received, but were
not being encrypted and decrypted. This resulted in no audio, or static.
Specifically, this occurred when '__rtp_recvfrom' was processing the handshake
sequence from the client to the server, and then 'ast_rtp_on_ice_complete'
gets called from another thread and clears the SSL object when calling the
'dtls_perform_setup' function. The timing had to be just right in the sense
that from the external SSL library perspective SSL initialization completed
(rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks
to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks
it is not finished, thus not completing 'dtls_srtp_setup'.
This patch removes calls to 'dtls_perform_setup', which clears the SSL object,
in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear
the underlying SSL object. If an ice candidate changes a full protocol level
renegotiation occurs. Also, in the case of bundled ICE candidates are reused
when a stream is added. So no real reason to have to clear, and reset in this
instance.
Also, this patch adds a bit of extra logging to aid in diagnosis of any future
problems.
ASTERISK-28742 #close
Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f
The change to res_config_odbc that allowed empty strings to be
returned to realtime consumers¹ causes a warning to be emitted when
loading MoH classes. So we need to treat an empty 'format' as if it
was not specified to avoid the warning.
ASTERISK-28735 #close
Reported by: Ross Beer
[1] https://gerrit.asterisk.org/c/asterisk/+/13722
Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5
Given a scenario where session refreshes occur close to
each other while another is finishing it was possible for
the session refreshes to occur out of order. It was
also possible for session refreshes to be delayed for
quite some time if a session refresh did not result in
a topology change.
For the out of order session refreshes the first session
refresh would be queued due to a transaction in progress.
This transaction would then finish. When finished a
separate task to process the delayed requests queue
would be queued for handling. A second refresh would
be requested internally before this delayed request
queued task was processed. As no transaction was in
progress this session refresh would be immediately
handled before the queued session refresh.
The code will now check if any delayed requests exist
before allowing a session refresh to immediately occur.
If any exist then the session refresh is queued.
For the delayed session refreshes if a session refresh
did not result in a topology change the attempt would
be immediately stopped and no other delayed requests would
be processed.
The code will now go through the entire delayed requests
queue until a delayed request results in a request
actually being sent.
ASTERISK-28730
Change-Id: Ied640280133871f77d3f332be62265e754605088
The code assumed that when the transport-cc feedback
function was called at least one packet will have been
received. In practice this isn't always true, so now
we just reschedule the sending and do nothing.
Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a
When res_config_odbc (and perhaps other realtime backends) reads a SQL
NULL from the database, it coalesces the value to the empty string
which prevents it from being returned to the realtime core.
However, if it instead reads the empty string from the database, it
needs a way to encode that fact without having the value omitted
entirely. It does this by changing the value to a string with a single
space. The realtime code in main/config.c recognizes this special case
and _turns the string back into the empty string_ before passing it to
realtime API consumers.
For all of this to work, we need to ensure that we actually pass the
single-space-string back to the realtime core, which is currently
failing because we are trimming the value before checking its
content. So instead we now special case the single-space-string case
so that empty values are returned properly.
ASTERISK-28719 #close
Reported by: EDV O-TON
Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de
Incrementing stasis_app_playback.media_index directly in our playback
loop means that when we reach the end of our playlist the index into
the vector will be outside of the bounds of the vector.
Instead use a temporary variable and only assign when we're sure that
we are in bounds.
ASTERISK-28713 #close
Reported by: Sébastien Duthil
Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928
Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.
ASTERISK-28714
Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a
Calling 'app_send' eventually calls the app's message handler. It's possible
for a handler to obtain a lock on another object, and then need/want to lock
the app object. If the caller of 'app_send' locks the app object prior to
calling then there's a potential for a deadlock, if another thread calls
'app_send' without locking.
This patch makes it so 'app_send' is not called with the app object locked in
the section of code doing such.
ASTERISK-28423 #close
Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27
The cleanup code in stasis shuts down applications if they are in a deactivated
state, and no longer have explicit subscriptions. When registering an app the
cleanup code was running before calling 'update'. When it should be executed
after 'update' since a call to register may re-activate the app. We don't want
it to shutdown before the 'update' otherwise the app won't be re-activated,
or registered.
This patch makes it so the cleanup code is executed post 'update'.
ASTERISK-28679 #close
Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b
We need to wait for the message sending callback to finish to know if
we succeeded or failed.
ASTERISK-25421 #close
Reported by: Dmitriy Serov
Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059
The change in 9b99ef50b5 updated the
syntax of the 'realtime update2' CLI command but did not correctly
update the calls to ast_update2_realtime().
The issue this addresses was originally opened because we aren't
allowing a SQL NULL to be set as part of the update, but this is a
limitation of the Realtime API and is not a bug.
Additionally, this patch:
* Corrects the example in the command documentation to reflect
'update2' instead of 'update.'
* Fixes the leading spacing of the command documentation.
* Checks that the required 'NULL' literal argument is present where we
expect it to be.
ASTERISK-21794 #close
Reported by: Cédric Bassaget
Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd
This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.
A description of the protocol can be found on the above referenced
GitHub page. A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.
ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.
ASTERISK-28484 #close
Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
Some body generators, such as dialog-info+xml, require storing state
information which is then conveyed in the NOTIFY request itself. Up
until now there was no way for such body generators to persist this
information.
Two new API calls have been added to allow body generators to set and
get persisted data. This data is persisted out alongside the normal
persistence information and allows the body generator to restore
state information or to simply use this for normal storage of state.
State is stored in the form of JSON and it is up to the body
generator to interpret this as needed.
The dialog-info+xml body generator has been updated to take advantage
of this to persist the version number.
ASTERISK-27759
Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de
Adds source port matching support when IP matching is used:
[example]
type = identify
match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444
If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.
ASTERISK-28639 #close
Reported by: Mitch Claborn
Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92
ast_sorcery_changeset_create() is not commutative and will fail to detect
differences between two variable lists depending on what changed, so switch to
ast_variable_lists_match().
ASTERISK-28492 #close
Reported by: Jean-Denis Girard
Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1
When TLS is in use, checking the readiness of the underlying FD is insufficient
for determining if there is data available to be read. So before polling the
FD, check if there is any buffered data in the TLS layer and use that first.
ASTERISK-28562 #close
Reported by: Robert Sutton
Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d
This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel'
operation in the Bridges REST API. When set, this flag avoids generating COLP
frames when the specified channels enter the bridge.
ASTERISK-28629
Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc
A previous review, 13174, made a change whereby on an incoming offer SDP
the pending topology was initialized to the configured. This caused a problem
for bundle with WebRTC where bundle could reference a stream that did not
actually exist if the configuration had both audio and video but the
offer SDP only contained audio.
This change undoes that review and instead fixes the original problem it
sought to solve by setting the state of created streams based on the
contents of the offer SDP. This way the stream state is not inactive
until negotiation later completes.
ASTERISK-28659
Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355
A previous patch:
Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39
made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.
This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).
This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.
The default is disabled keeping with the old behavior.
ASTERISK-28660
Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a
In Asterisk 16+, there are a few places in ast_rtp_read where we've
allocated a frame list but return a null frame instead of the list.
In these cases, any frames left in the list won't be freed. In the
vast majority of the cases, the list is empty when we return so
there's nothing to free but there have been leaks reported in the
wild that can be traced back to frames left in the list before
returning.
The escape paths now all have logic to free frames left in the
list.
ASTERISK-28609
Reported by: Ted G
Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a
RFC3261 Section 10 "Registrations", specifically paragraph
"10.2.4: Refreshing Bindings", states that a user agent compares
each contact address (in a 200 REGISTER response) to see if it
created the contact. If the Asterisk endpoint has the
rewrite_contact option set however, the contact host and port sent
back in the 200 response will be the rewritten one and not the
one sent by the user agent. This prevents the user agent from
matching its own contact. Some user agents get very upset when
this happens and will not consider the registration successful.
While this is rare, it is acceptable behavior especially if more
than 1 user agent is allowed to register to a single endpoint/aor.
This commit updates res_pjsip_nat (where rewrite_contact is
implemented) to store the original incoming Contact header in
a new "x-ast-orig-host" URI parameter before rewriting it, and to
restore the original host and port to the Contact headers in the
outgoing response.
This is only done if the request is a REGISTER and rewrite_contact
is enabled.
pjsip_message_filter was also updated to ensure that if a request
comes in with any existing x-ast-* URI parameters, we remove them
so they don't conflict. Asterisk will never send a request
with those headers in it but someone might just decide to add them
to a request they craft and send to Asterisk.
NOTE: If a device changes its contact address and registers again,
it's a NEW registration. If the device didn't unregister the
original registration then all existing behavior based
on aor/remove_existing and aor/max_contacts apply.
ASTERISK-28502
Reported-by: Ross Beer
Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e
The simple fix here is simply to NULL out username and password after we call
ast_free on them. Unfortunately, I noticed that we weren't checking for
allocation failures for username and password, and adding those checks made
things noisy and cumbersome.
So instead we partially rollback the recent LGTM patch, and move the alloca
calls into find_aor_name().
ASTERISK-28641 #close
Reported by: Ross Beer
Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2
We're appropriately sizing the id_domain_alias buffer, but then copying the data
into the id_domain one. We were then using the uninitialized id_domain_alias
buffer we just allocated.
This is ASTERISK~28641 adjacent, but significant enough to warrant its own
patch.
Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9
We need to copy the endpoint name before we call ao2_cleanup() on it,
otherwise we might try to access memory that has been reclaimed.
ASTERISK-28445 #close
Reported by: Bernhard Schmidt
Change-Id: I404b952608aa606e0babd3c4108346721fb726b3
Joshua C. Colp
Nickolay Shmyrev
Ben Ford
George Joseph
sungtae kim
Kevin Harwell
Università di Bologna - CESIA VoIP
Moises Silva
Walter Doekes
sungtae kim
Pirmin Walthert
Alexander Traud
Nicholas John Koch
Roger James
Guido Falsi
DanielYK
bernard merindol
Jaco Kroon
Sean Bright
Torrey Searle
Michael Neuhauser
Rodrigo Ramírez Norambuena
lvl
Friendly Automation
Seán C McCord
Jean Aunis
Joshua Colp