According to:
{
"request" : "join",
"ptype" : "subscriber",
"room" : <unique ID of the room to subscribe in>,
"use_msid" : <whether subscriptions should include an msid that references the publisher; false by default>,
"autoupdate" : <whether a new SDP offer is sent automatically when a subscribed publisher leaves; true by default>,
"private_id" : <unique ID of the publisher that originated this request; optional, unless mandated by the room configuration>,
"streams" : [
{
"feed" : <unique ID of publisher owning the stream to subscribe to>,
"mid" : "<unique mid of the publisher stream to subscribe to; optional>"
"crossrefid" : "<id to map this subscription with entries in streams list; optional>"
// Optionally, simulcast or SVC targets (defaults if missing)
},
// Other streams to subscribe to
]
}
{
"videoroom" : "attached",
"room" : <room ID>,
"streams" : [
{
"mindex" : <unique m-index of this stream>,
"mid" : "<unique mid of this stream>",
"type" : "<type of this stream's media (audio|video|data)>",
"feed_id" : <unique ID of the publisher originating this stream>,
"feed_mid" : "<unique mid of this publisher's stream>",
"feed_display" : "<display name of this publisher, if any>",
"send" : <true|false; whether we configured the stream to relay media>,
"ready" : <true|false; whether this stream is ready to start sending media (will be false at the beginning)>
},
// Other streams in the subscription, if any
]
}
Change-Id: Ieb38d4f562686283457a963334056b27927be974
Apparently Janus allows a single handle to be both controller and
publisher, and possibly also both publisher and subscriber. Remove the
limitation that forces one handle to only take one role.
Change-Id: I22c20981a213b84d06dad07b1ff634bfd024fd91
This is a new option flag, which provides a possiblity
to select specific crypto suite(s) for the offerer from
the given list of crypto suites received in the offer.
This will be used later on, when processing an answer from
the recipient and generating an answer to be sent out towards offerer.
Furthermore, this is being decided not when the answer is processed,
but already when the offer is processed.
Flag usage example:
`SDES-offerer_pref:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;`
Change-Id: I2b22b38347d24f27331482e18b92305fbadb2520
We're already determining the destination media section in
media_player_play_init() (looking for audio media), so it makes sense to
determine the payload type there as well.
Change-Id: I33d4a15005c2146864f74663fe8de2893454345b
Split out the logic to determine the destination payload type into a
separate functions. This makes it possible to supply a different, or
pre-determined, payload type instead.
Change-Id: I9b67b29cafc0c6ce4e18eede64dea3d1973f8b63
It makes more sense to set up the codec handler from the init function
rather than from the function reading the media packet.
Change-Id: I0bea3d0fc0669a54f93b2598251df69274950365
Provide the targets for the two /bin/ components.
Adapt .install debhelper files.
Adapt iptables module install location to use the environment variable.
Change-Id: I963feba5f60f53773e497121d8947e7b4997d687
… if it does not exist. Likewise in iptables-extension/Makefile .
daemon/rtpengine.pod: spell what MOS means
MAX_SESSIONS in config file does not work, it must be max-sessions.
https://github.com/sipwise/rtpengine/pull/1589
1872 /* first add those mentioned in the order list,
1873 * but only, if they were previously generated/added to the sdes_out */
>>> CID 1530346: Control flow issues (DEADCODE)
>>> Execution cannot reach the expression "NULL" inside this statement: "l = (cpq_order ? cpq_order-...".
1874 for (GList *l = cpq_order ? cpq_order->head : NULL; l; l = l->next)
Change-Id: I58f84ba82d215a9eb6cbd97548e8e9e8a954bdc6
This is a new option flag, which provides the ordered list,
in which to add crypto suites into the SDP body.
Right now they're always added in the order given in the source code.
Flag usage example:
`SDES-order:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;AES_192_CM_HMAC_SHA1_80;`
This means — those listed SDES crypto suites will be added
into the generated SDP body at the top of crypto suites list, in the given order.
But, each of them is added, only if it is about to be added/generated.
In other words, the `SDES-order:` flag itself doesn't add crypto suites,
it just affects the order of those suites to be added.
And the rest of non-mentioned suites, which are also to be added,
will be appended after those given, in the free manner of ordering.
Important thing to remember - it doesn't change the crypto suite tag
for the recipient, even though changing the order of them.
Additionally.
This flag does not contradict with `SDES-nonew`, `SDES-only-` and `SDES-no-` flags.
It just orders the list of crypto suites already prepared to be sent out.
Change-Id: I0fec54f9e2f3cd4913e905e8afe825712f82d1ae
A new function dedicated to SDES crypto suites comparison.
It compares the crypto suites using a name in 'str' format.
Recommended to be used in combination with:
g_queue_find_custom() or g_list_find_custom()
Change-Id: I08ff6d3304f74d29154110caa472618478ca1837
New helper func: call_ng_flags_str_q_multi()
It parses given flags delimited by a separator, and stores
each of them in the given GQueue.
By default delimiter is assumed as: ';'.
Change-Id: I1e29bf3d852868e2bc4d98b1775f70b38f61054a
This type is used as const everywhere except internally, so make it part
of the typedef for brevity.
Change-Id: Ic4afe037b392239a991d5380c6708903011da29e
A new function dedicated to SDES crypto suites policy checks
has been introduced: 'crypto_params_sdes_check_limitations()'.
Use it to decrease an amount of repeating code blocks
related to SDES checks.
Change-Id: I0ac242a63107a9f3a41f95a57e3d3675645ac18d
Add a new flag to only accept these individual crypto suites
and none of the others.
For example, `SDES-only-NULL_HMAC_SHA1_32`
would only accept the crypto suite `NULL_HMAC_SHA1_32` for
the offer being generated.
This also takes precedence over the `SDES-no-` flag(s),
if used together, so the `SDES-no` will be not taken into account.
This has two effects:
- if a given crypto suite was present in a received offer,
it will be kept, so will be present in the outgoing offer; and
- if a given crypto suite was not present in the received offer,
it will be added to it. The rest, which is not mentioned,
will be dropped/not added.
Flag name: 'SDES-only-<crypto name>'
Additionally: add another new flag 'SDES-nonew'.
It will not add any new crypto suites into the offer.
It takes precedence over the `SDES-no` and `SDES-only` flags,
if used in combination.
Change-Id: Ic4fa03957ee3d4d24b0c4f3fd003eada05f49b0b
1) Read dtxb->start while the lock is held (not strictly necessary as
the read should be atomic anyway)
2) Expect that dtxb->start can be larger than rtpe_now.tv
The latter can happen as `rtpe_now` in a timer thread is faked to be
exactly the time when the timer was supposed to run, and not the actual
current time, which means that a newly added packet can have a later
time stamp than the "now" the timer thread is using.
Change-Id: I48fd7f78af97c6d5b802e5151d69855a90f4032d
Instead of going through ffmpeg to en/decode Opus, use libopus directly,
which allows us to benefit from additional features that aren't
available when going through ffmpeg.
Change-Id: I017c276cfa9755cefe95c8da26691446b718d4c8
Some codecs (e.g. Opus) can natively encode audio with various clock
rates without producing an output that is locked to that clock rate and
without requiring resampling the input. Add an appropriate callback
function and adapt tests.
Change-Id: Id788c4d4c05e20f93cce7e910f9f265b381cbe34
This makes it possible to have codecs running at variable clock rates
that differ from their RTP clock rates.
Change-Id: Ia2f5effb82eefe8c3028573ba0a6697da28473b1
Instead of just having an integer multiplier, support a fractional
factor. This allows us to have the RTP clock rate run faster than the
audio clock rate, and not just slower by an integer factor.
Change-Id: I7681cf369c43d8424ca2d2ebeffe932595d271ec
Unmark a monologue that has been scheduled for deletion when it's
associated with another one, which happens during offer/answer.
Fixes a regression from 53dbef7e1 which removed this logic from
__monologue_unkernelize with re-adding it elsewhere.
Change-Id: I037162e91fec42631680f7767f58b172fd6e04db
Add a flag to force increasing the SDP version,
even if the SDP hasn't been changed.
And cover it with tests.
Flag name: 'force-increment-sdp-ver'
Additionally fix the name of the 'sdp-version' flag
in the 'rtpengine-ng-client' tool.
Change-Id: I466792668b0cd313b5e21b248dd14cd599333cbd
`call_get_monologue_new` is supposed to always return a full dialogue,
but an error in invocation (using a from-tag that doesn't belong to an
offer/answer) can lead to the second half being unset. Return an error
in this case.
Change-Id: I84b21ff5e5c0403fc07cae83fee206705ecff8b3
Use the new `associated_tags` table to determine which tags are
associated with which. Iterate the associations between tags in a
tree-like manner and do this at the moment the `delete` command is
received. Break up the `associated_tags` links at this time, and
determine which tags would be left dangling and mark all of these for
deletion. If no tags are left after this process, mark the entire call
for deletion.
The previous approach was cumbersome and prone to errors. Using tag
names and branch names to determine which tags are associated with which
is a pointless hurdle, and using a table of associations that is
explicitly kept for this purpose is a much cleaner approach. Also
postponing the decision about which tags to delete until the time the
deletion actually happens can lead to tags not being deleted, when they
really should be (e.g. A -> B, delete A, A -> C).
Change-Id: I03ae57d0a2117ecd721372c1a49468fc34dd630c
Keep track which tags (monologues) were created together as part of an
offer/answer exchange with a separate hash table, regardless of whether
these monologues actually have tagged names or are just nameless
branches.
Change-Id: I60aa114c8caf6ecdff4705e3399f60190d04dda6
During offer/answer, we remove existing subscriptions before
re-establishing the new set of subscriptions. But we must also remove
the subscriptions of previous subscribers in order to keep the lists of
subscriptions and subscribers correctly reciprocal.
Change-Id: I7d39fa59892159f41033ae6a40e7cb51d861b12e
Reset the iteration counter for the frequency list when codec handlers
are set up, which happens while setting a new block mode. This makes
sure that a newly set list of frequencies is iterated starting from the
top.
Change-Id: I7ab4e28a1a998c11e38e26cf5b1f17f01299d5ad
Support multiple tone frequencies for DTMF-security=tone to enable
audibly distinguishing multiple consecutive DTMF events from one
another.
Change-Id: I6fa33a5768aae198220d0b0cc4c53308c5661a52
The `dtmf_send` event might not have a matching entry while handling a
received DTMF event. Move bookkeeping of the random output event into
`dtmf_recv`.
Change-Id: I5507b0b7f5eca6e29cbcaccc905ab25249b22aa3
Usually supplemental RTP types (DTMF) are listed after the primary audio
codecs. In the case of the order being reversed, fix `single-codec` so
that it doesn't strip the actual audio codec that is listed after the
DTMF type.
Change-Id: I1b03b89e31bebf4de303b643dcf08d2ffb90ebaf
In some cases it's possible that some packets still arrive in userspace
immediately after a stream has been pushed to the kernel, for example if
some packets are already in the queue or if there is some processing
delay (e.g. writing to Redis). Allow for a short delay before counting a
stream as userspace if it has been pushed to the kernel.
Change-Id: I55a6e255868c8c2a9e93355a4aa2287f07b3748d
Based on the information gotten from Richard Fuchs
document the main objects in the code, to let the code be more
understandable for other code readers.
Mainly documented:
- call
- call_monologue
- call_subscription
- call_media
- packet_stream
- stream_fd
- sink_handler
- rtpe_callhash / rtpe_callhash_lock
Change-Id: I0cf122bea2d9c3f198b48da134a70301564ff1f9
On bookworm and later, libasan reports a false positive in combination
with pthread_cleanup_push() (see [1]). Work around this by not using the
thread cleanup handler when running the asan build, and instead use a
shorter thread sleep time.
[1] https://gcc.gnu.org/bugzilla//show_bug.cgi?id=82109
Change-Id: Ieffdc0b13f470445f1f8e1d2448c6af6d8dd68e0
Only skip forwarding unknown payload types in the kernel module if
there's reason to skip them, i.e. when we know that certain payload
types must be handled in userspace. Use an explicit flag to signal this
to the kernel module instead of implicitly doing it for anything that is
RTP.
Change-Id: I655317afe64a27252bf7b8be6c78418db2e1ccef
Legacy UDP/TCP control protocols don't provide information about RTP
payload types, therefore don't pretend that we know that this is RTP.
Setting the RTP flag without knowing the payload types has the undesired
side effect that unknown payload types (all of them) would not be
handled by the kernel module.
Change-Id: I5882f777a5912b912ec7c870f21c77aac8127600
Instead of just leaving the transport protocol unset when we know we're
not supposed to be aware of the protocol, add a special entry to
suppress the pointless warning message.
Change-Id: I228c2f1652320627f974d9d7bcb0b1345adce2be
Similar to I7fd298669a4fd69e62edc4760ffd92eff074707b but for
codec-accept. This prefers accepting codecs that match the given
definiton exactly, but also accepts compatible codecs if no exact match
is found.
Change-Id: I52a2ab6e75f7334a85a8ef059a0d6703a07e4b7f
Legacy protocols don't set sp->num_ports. Use a sensible default for
this case.
Probably a regression from 2d2d7665b
Change-Id: Idcbc477a68b6db70a91a5d082736ac642c50ab15
When adding codecs to the SDP for potential transcoding, use more
sophisticated logic to determine whether the desired codec is already
present. Previously only the raw RTP payload string was used, which
doesn't take codec-specific format options into account. The new
matching function will now also match against existing codecs with
slightly different but compatible a=fmtp options, but will not match
against those with incompatible options.
Change-Id: I7fd298669a4fd69e62edc4760ffd92eff074707b
Convenience function to get a list of all codecs that match a given
definition, including a possible exact match.
Change-Id: I2b7d49fb320ad1725b348f7d29be450e5fea925b
Make these functions return the newly created payload type object so
that it can be manipulated directly without having to do a lookup.
Change-Id: Ib035cf7d61c61d0de6d5f87dfd497d17ba28ca5f
Parse out the format parameters for a payload type object when the codec
definition is loaded. Parsing it out is done only once, and if done
early the parsed contents can be used for format checking, comparing,
matching, etc.
Change-Id: I2856e5a8d4d76e7e1e648eff4a91fc51d0ba20c2
Split up this function into two parts so that they can be used
individually. Eliminate the special 0x1 return value as this isn't used
anywhere else. Move the error reporting into
codec_make_payload_type_sup() where it belongs.
Change-Id: Id2060272d87596a3db32d8776930a98b0f226b3b
Introduce an additional function for codec matching. Different functions
match different parts of an RTP payload type object and the referenced
codecs.
Change-Id: I2e488eaa7f69a55322db748fd40c8d1195e38605
We have no use for -1/0/1 return values. Change the return type to bool
to make things more clear.
Change-Id: Iedf1d8278c6dfddddb328ce7b3b1dbae132a39b7
Parsing out the a=fmtp string has been left up to the codec init
function until now, with the values that resulted from the parsing being
stored only within the codec. Convert this to an explicit method to
parse the a=fmtp string, and introduce a dedicated struct to store the
resulting values.
Functionally this change is a no-op.
Change-Id: Ia84e26d632ed5209b4439fd82c1e4e38850fd024
commit 025f56212d
Author: Andreas Granig <andreas@granig.com>
Date: Tue Oct 4 14:23:01 2022 +0200
Document the mqtt-tls-alpn option
commit e6cc320d19
Author: Andreas Granig <andreas@granig.com>
Date: Mon Oct 3 21:41:14 2022 +0200
Add TLS ALPN option to be set when connecting MQTT
This is required to be set to "mqtt" for instance when connecting
to the AWS IoT Core data endpoint at port 443 to indicate we're
sending MQTT, because in that case websocket and mqtt shares the same
port for whatever reason.
Change-Id: I6a391e815411b178187ef42aa009e45853d1c388
The packet must be decrypted first before RTP padding can be considered,
as the padding count is part of the encrypted payload as well.
Change-Id: I6aecff636efd420401856bb8110b3d784f989179
This distinguishes `to-label` from `set-label` for media blocking
methods, when previously they were synonymous.
Upgrade sink determination to list at the same time.
Change-Id: I5b35c78f2f307867b51b5376d5a6afbd79128d99
This makes it possible to have different media silencing options for
different outputs. Functionally this commit alone is a no-op.
Change-Id: I967c3e07ea4645bb49ccb76db12d51ded2d72f06
Make kernelize_one eturn the list of sorted payload types back to the
calling function instead of only keeping it temporarily. This makes it
possible to re-use the list in repeated invocations of the same
function.
Change-Id: I696b4d033715bb60c80c8b932b80d558f364a5c5
Don't keep the flag for "no kernel support" set indefinitely as
conditions might change, re-enabling kernel support at a later time.
Change-Id: Id8f456b653f1ea3d6a9db1ff3a0800d3cca8fcc5
Create a dedicated struct to hold certain attributes shared by both sink
handlers and media subscriptions, as a preparation to simplify handling
these attributs.
Change-Id: I866159c33ed6d6a2873d2cf68c4906ea705d253e
This makes it possible to refactor and simplify the interface functions,
as pointers and offsets can't be utilised with bit fields.
Change-Id: I70f1ac0eca7d2ccf8e8d5f5794580163f3f5b7ad
If an offer is going to a side that is already known to support
rtcp-mux, don't try to offer a fallback RTCP port. Do this by pretending
that rtcp-mux=require is set, which leads to the correct behaviour. This
is needed because the RTCP fallback port might already be closed, so
trying to include it in the offer would require opening a new set of
ports, which is undesirable.
closes#1494
Change-Id: I550bec08379c799cb7dd090a70d090ae47462467
Newer libwebsockets versions seem to use a longer internal timeout, so
an explicit "interrupt" is needed during shutdown to prevent a long wait
time.
Change-Id: I8f28ef658169178e35b40dd44520fbd7c812b590
Zero SSRC are technically invalid, but the code accepts them as valid
and therefore sets up things like crypto contexts for zero SSRCs.
Consequently it's possible to have a zero SSRC first in the list of
SSRCs and other valid SSRCs later in the list, which means we can't use
the presence of a non-zero SSRC first in the list as a flag to determine
whether SSRC tracking is in use or not. Use an explicit flag instead.
Change-Id: I88736e5d6b0f66c58f8d675137231760951e7610
Don't increase the index pointer before using it. This fixes slots 1 and
2 being filled before slot 0, which ends up filling slots 0 and 2 first
(due to swapping with slot 0), leaving slot 1 empty.
Change-Id: If34cf561fcb153edde6408252c3286c8c80991bc
Keep a running lifetime total of all "gauge" type metrics. Also track
the square of the sums of all "gauge" type metrics in order to determine
the standard deviation.
Change-Id: I23f60774a6421636f1a913674c7d1b54a1c5f702
Starting with release 5.1, ffmpeg obsoletes the `channels` and
`channel_layout` fields, replacing them with a unified `ch_layout`
struct of type AVChannelLayout. Add wrapper defines into the
compile-time build-test headers fix_frame_channel_layout*.h to
accommodate both new and old versions.
Change-Id: I3d43b85dc3140155a61b1cf2269cda166ad88e9a
Keyspace notifications are set up before existing calls are restored
(restore_thread). Therefore the following scenario is possible:
NOTIF THREAD: receives SET, creates call
RESTORE THREAD: executes KEYS *
NOTIF THREAD: receives another SET:
NOTIF THREAD: does call_destroy(), which:
adds ports to late-release list
RESTORE THREAD: comes across call ID, does GET
RESTORE THREAD: creates new call
RESTORE THREAD: wants to allocate ports, but they're still in use
NOTIF THREAD: now does release_closed_sockets()
Use a multi-A multi-B lock to protect the two code sections from each
other and make sure that notification threads have always released all
ports before a restore thread attempts to allocate any.
Reported in #1503
Change-Id: I322062488e2ce3515c5a3e6609a5700830ac1fd4
To prevent a race condition that might miss updates about call info, set
up the Redis keyspace notifications first and then run loop to restore
calls from the existing data.
closes#1503
Change-Id: I6afa4c50fe0a34c602063fc2f45b2ee38133cf1e
Allow override the bitrate if one is already set, as opus_init()
initially sets a default bitrate, which is then overridden by codec-set.
Change-Id: Id0253d23a8f5de977e30d296872ea1df01fdbfbc
The PT tracker doesn't distinguish between audio/media types and
supplemental types, so in order not to break DTMF handling we must take
all combinations of primary (audio/media) types and supplemental types
as both input types and handler types into account.
Fix-up for 74075f6396
Fix-up for I57e1278e4fad157083d9526d4829f2940581687f
closes#1508
possibly also #1504
Change-Id: If7b242def2d35fbed14b11d204ea328b8bfe5d79
To safeguard against non-refcounted objects being left over in a log
info piece (e.g. a string on the stack), add this new function to pop
pieces from the stack until the desired one is removed. This is needed
in case of a unpaired log_info_* without a matching log_info_pop.
closes#1511
Change-Id: I689de14d034df779521dfdf59f923fdbf7fabc9b
To safeguard against leftover log info pieces, add additional resets
within loops that might run repeatedly.
Relevant to #1511
Change-Id: I875f1683b7dc8cee359469e8062c08c3c3e48a9d
The codec answer routine resets the codec storage and so also resets the
clock rate tracker for "touched" codecs. This leads to all codecs seen
as "not touched" in the answer routine, which in turn leads to
supplemental codecs present in the answer SDP that should not be there.
Use the "for transcoding" flag for previously present codecs to retain
the "touched" status across the codec answer routine.
Change-Id: Idc4624606f7f10d7983e22ddf856432b07421157
When doing the initial answer, the packet_stream endpoint port isn't
filled in yet. Use the stream_params port instead to test for rejected
streams.
closes#1499
Change-Id: I8f315d95521f874fb8c5e6222263d017800b5fc9
When ports are closed early (while the call is still running), we must
first update a slave rtpengine with this new information (that these
ports are now closed) before actually releasing the ports ourselves. Not
doing so leads to a race condition where the master instance re-uses a
port that was just closed before the slave instance knows about the port
being closed.
We implement this using a thread-local list to keep track of ports that
were released while processing a control message, and process this list
to actually close the ports only after Redis has been updated.
Additional calls to the function to close the ports are placed in
strategic locations to make sure this is triggered in every code path.
closes#1495
Change-Id: I803f4594f30ca315da0b84c6e76893f54ca3a7c9
This prevents empty mixed output files from being created when mixed
output is enabled in the config but recording isn't active for that
call.
Change-Id: I66ead89dc8a7ea80b81164b3e24d997b0df5f37e
DTX and delay buffers and their timers are shut down during the codec
negotiation phase, which also happens for the offer side while
processing an answer. If the codec negotiation routine determines that
the existing codec handlers can be kept intact, we must restart the DTX
and delay buffers that have previously been shut down.
Buffer objects are never freed during a shutdown, therefore we simply
need to restore the contained references to indicate that these buffers
are active again.
closes#1481
Change-Id: I57181ba1655fd781a7c543ee31aa67fd179ba89b
This eliminates a spurious false warning log message for rejected
streams that use a dummy payload type
Change-Id: Id628cafb8d7c4ea576cd01ff35f5dd9cd2151280
Since we're already doing the full parsing of the request flags, use the
same function to parse all required flags
Change-Id: I0880ccbbbc36eae7b172440ce51afc1c544583a1
Instead of always generating a new ICE foundation string for every
learned peer-reflexive candidate, try to use the same foundation string
for candidates that belong together. We use the priority number plus the
component ID for this to see if we've learned a candidate with a fitting
priority number before. If we have then re-use the same ICE foundation
string. This allows ICE to complete with only learned prflx candidates
and without (or before) re-invite to communicate the correct ICE
foundations.
Change-Id: I74bde6ef22a164df57d0b77cbaef34e4a499da72
`flags` being NULL is not really a use case we have right now, but it
might be in the future. Make sure we don't crash.
Change-Id: I4400553fc0a665d94f2e1cdced855250b46d88a4
Warned-by: coverity
Initialising the other members of this struct is not really necessary as
they're not used in the hash lookup. But let's do it anyway.
Change-Id: Ia7cf982fe91e9c4d273b1fc2d2ee8b19ce345a13
Warned-by: coverity
If HMAC() fails, the value of the output string would be left
uninitialised. Handle this case.
Change-Id: I79fc3d03237ae4a5924e59f749d6818db7bf8ab2
Warned-by: coverity
Since we're creating a dummy sfd to hold the SRTCP context when we don't
have an actual RTCP port, we must make sure to remember and re-use this
dummy sfd during a re-invite. Otherwise we end up creating a duplicate
dummy sfd, which is detected as a different sfd and thus triggers an ICE
restart.
Change-Id: Iadc91e163bd15a3cd5f57656b52941724c920143
Explicitly copy SDP up to the format list before printing it out. This
preserves broken input SDP.
closes#1461
Change-Id: I839a200f159f25854c86add244571a948e2c90cf
Setting the mux flag when rtcp-mux is given is fine, but we must still
provide an RTCP endpoint in case rtcp-mux ends up not being used, either
through an implicit RTCP endpoint or through a=rtcp.
relevant to #1443
Change-Id: I0710a50c31974f5e06bd94b47076a272bcca7a43
(cherry picked from commit e3951449ed)
There's no need to open ports on non-primary interfaces if ICE is not in
use as these ports will not be used or seen by anyone.
This mostly obsoletes the `save-interface-ports` config option, with the
exception of ICE advertised by the offerer. We currently have no option
to reject ICE from the offerer during the offer phase, so ports would
always be opened on that side.
Relevant to #1164 and 001abe5
Change-Id: I43df70bc0ec49b81f63aec97c776e48617b2acfd
This enables the same behaviour towards the offerer when rtcp-mux=demux
or =accept is used, as we have towards the answerer when
rtcp-mux=require is used.
Change-Id: I56a1cea84efce0c2db1b58c500629d0e54d582f4
Special handling for codec lists that were received as part of an
answer: If the list includes a codec that was not offered, ignore that
codec. This prevents transcoders from being set up that were not
requested.
This brought to light some tests that were actually broken.
Change-Id: Iac71056ec5e10b5de5567917974f2c4e0261eb0c
We must now hold the master lock for reads from the socket as the socket
may get closed after the poller has already fired an event for it.
Change-Id: I1ab4b38f09988e8569a70c449de17c208ef2aa96
If DTLS is rejected in an answer via `DTLS=off` we must forget that DTLS
was previously offered, as otherwise a re-invite would detect the
fingerprint as changed if the re-invite doesn't offer DTLS again. We
also make sure DTLS is shut down if during stream init DTLS is not
given, when it was present before.
Change-Id: I48ee6f0ec5ec02f558a6799951552ea2272d0e96
All crypto suites except AEAD have an explicit packet authentication
stage. If authentication fails for a packet, we take some guesses about
a ROC mismatch and see if authentication can succeed with a different
ROC. If a working ROC is found, our tracked ROC is updated and
decryption proceeds.
AEAD doesn't have an explicit authentication stage and authentication is
performed implicitly by the decryption engine, which simply returns a
decryption error if the authentication fails. We must therefore add the
same ROC guessing logic at this step for AEAD.
Change-Id: Ic1a70daa667e23976b74d2303c823b8d8c7bcb2b
This is useful for functions which are used both from a timer and from
other callers. These functions would reset the logging context at their
end to free the reference held by the logging context, which would
wrongly reset the logging context when the same function was called from
a different code path. Using a stack with push/pop semantics makes it
safe to use these functions from any code path.
Additionally introduce an explicit reset function that clears the entire
stack regardless of context. This reset function is called at the end of
every work iteration in every worker thread, just in case not everything
was popped from the stack.
Change-Id: I0e2c142b95806b26473c65a882737e39d161d24d
If the config only lists a port for the HTTP/WS bindings then we must
not try to create both a v4 and a v6 binding on that port as
libwebsockets handles the 4/6 mapping internally. In this case we make
sure to only create the v6 binding.
Further requirement for #1432
Change-Id: I9bf7ec5c041d0b5d4a22d507d993b85e2d4d3155
Add an explicit test to see if libwebsockets has been compiled with
support for IPv6. If it hasn't then we don't try to create v6 bindings.
Closes#1432
Change-Id: I6902f5b4203aa09cb28a8edb46f97b339677ed75
commit a2e5cfb8e5
Author: Razvan Crainea <razvan@opensips.org>
Date: Thu Jan 13 16:16:19 2022 +0200
Add tests for subscribe requests on paused media
commit fa58596a9f
Author: Razvan Crainea <razvan@opensips.org>
Date: Wed Jan 12 22:01:27 2022 +0200
Swap media direction check for `subscribe request`
as @rfuchs mentioned in his review, the SEND/RECV media flags are set
according to rtpengine's perspective, not the media flow's one.
commit e1e9a157c0
Author: Razvan Crainea <razvan@opensips.org>
Date: Wed Jan 12 19:27:42 2022 +0200
Fix `subscribe request` SDP media direction
When building the SDP for a `subscribe request` command, take into
consideration the media direction of the source stream - if stream is
`recvonly`, then we do not have anything to send, thus the direction
should be advertised as `inactive`, rather than `sendonly`.
Change-Id: I2d78bbec8ad584774f3c90f0ce5cca42f57f7b0f
Instead of having each thread sleep only a little while and then
periodically check for the shutdown flag, make them sleep longer and use
pthread_cancel() to interrupt the sleep during a shutdown in the
designated break points.
Change-Id: I13f1872a0176697e064ceef4062db6ca6ccf7a0e
Handling of dual stack v4/v6 was previously done by the individual
listener objects for INADDR_ANY listening addresses. If listening on
INADDR_ANY was requested, then each listener would create two instances,
one for IPv4 and one for IPv6. This works fine for INADDR_ANY but fails
for listening on host names that resolve to multiple addresses, such as
`localhost`.
Solve this by relieving the listener objects from handling this and
instead handle it in the code setting up the listeners. If a host name
resolves to multiple addresses, then set up multiple listeners (up to
two supported currently). This allows us to listen on `localhost` by
default and have both 127.0.0.1 and ::1 active. INADDR_ANY is handled
specially by also setting up :: in that case.
Change-Id: I2a1e1d7090d7d23863c7a9bb1e89b85ad2ea44f4
Allow usage of "any" as interface config option to configure any and all
locally present network address, except loopback. This allows us to ship
a working default config file.
Change-Id: Ic13efd5f668e3bb317948b226c5700331f95a708
Needed to be able to set graphite socket timeout.
Useful when one wants rtpengine to force the graphite connection
to fail faster, in case graphite server gets filtered while
connection is ongoing.
Supplemental codecs such as DTMF use static timestamps while the event
is ongoing, leading to a TS jump when the RTP flow changes back to
audio. The sequencer needs to be aware of this so it doesn't mistakenly
see the next audio packet as overdue and starts to process it
prematurely.
Change-Id: I2faea9aceec21fc04920f6c3c94141725383379f
... for scheduling output RTP packets. This is mostly relevant for DTMF
packets which don't have an associated encoder when being forwarded.
Change-Id: I56ee94a9ac7f42cc65eec0703bf042065687e43f
With multiple media subscriptions, codec handlers are called
consecutively, once for each forwarding chain, leading to DTMF events
reported multiple times. The DTMF trigger must therefore keep track of
the state in the upper media object, not in the codec handlers.
Change-Id: I9ceaf406e093f25b7c037a325a0f2a7a91954922
Some functions (packet_dtmf in particular) called from the sequencer
depend on upper-level locking, so make sure this happens even if we're
bypassing the sequencer and do passthrough.
Change-Id: I6c729c3ba8075736fd614b8c06e3415b9c9e5ca7
SSRC entries might be present for the same SSRC in multiple contexts,
but only one of them will hold the actual stats. Don't create output
SSRC entries unless we know they won't be empty, as otherwise we won't
be able to create the actual SSRC entries (with stats) later on as they
dict key will already exist.
Change-Id: I54e263a17e14869ebb98456963f8ca75d11e9a89
This is useful because we log to stderr, which technically would allow
unlimited log line length, but this is in fact turned over to syslog,
which truncates log lines that are too long.
Change-Id: Iee8994842335ab1cf94941c14eced01e29120bc9
Not only check for the presence of a sink, but also check for a sink FD.
Treat a sink without an FD as if there is no sink.
Closes#1401
Change-Id: I04c0be33f8cae39399674ca0a87185a729daa843
Flag a socket with an error strike when packets are received too fast,
and refuse processing once too many strikes have occurred. This should
prevent forwarding loops from taking down the system.
Change-Id: Idc574f2f1dbbcb156efc37a80e903dc4e60ef1b1
libcs are implementing changes to fix the year 2038 issue on 32 bit
platforms (see [1]). musl libc already went ahead and implemented it,
starting with musl-1.2.0 (see [2]).
This commit adds a new definition to lib/loglib.h:
TIME_T_INT_FMT
If __USE_TIME_BITS64 is defined (by a time64 libc, see [1]), it's set to
the proper conversions for type int64_t, PRId64. If __USE_TIME_BITS64 is
not defined, the status quo remains unchanged ("%ld" is used).
The new definition is used in the different parts of rtpengine, where
appropriate.
Note: Richard confirmed that the "%u" format in daemon/cdr.c is not
needed, so this gets swept under the rug.
These changes get rid of the new warnings that appeared with musl-1.2.0.
Below an example warning:
In file included from ./log.h:6,
from ../include/obj.h:94,
from ../include/media_socket.h:9,
from ../include/call.h:26,
from ../include/redis.h:15,
from redis.c:1:
redis.c: In function 'redis_check_conn':
../lib/loglib.h:56:30: warning: format '%ld' expects argument of type 'long int', but argument 5 has type 'time_t' {aka 'long long int'} [-Wformat=]
56 | __ilog(prio, "[%s] " fmt, log_level_names[system], ##__VA_ARGS__); \
| ^~~~~~~
../lib/loglib.h:64:39: note: in expansion of macro 'ilogsn'
64 | #define ilogs(system, prio, fmt, ...) ilogsn(log_level_index_ ## system, prio, fmt, ##__VA_ARGS__)
| ^~~~~~
../lib/loglib.h:63:30: note: in expansion of macro 'ilogs'
63 | #define ilog(prio, fmt, ...) ilogs(core, prio, fmt, ##__VA_ARGS__)
| ^~~~~
redis.c:887:17: note: in expansion of macro 'ilog'
887 | ilog(LOG_WARNING, "Redis server %s is disabled. Don't try RE-Establishing for %ld more seconds",
| ^~~~
[1] https://sourceware.org/glibc/wiki/Y2038ProofnessDesign
[2] https://musl.libc.org/time64.html
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
This makes it possible to add new streams without specifying the
direction/interface again.
Reported in #1366
Change-Id: I8f320ecbe72f123d755ba80370de9c40960eb0f0
Richard Fuchs
Lasse Bang Mikkelsen
Stefan Mititelu
Donat Zenichev
Дилян Палаузов
Damir Nedžibović
Dup4
Andreas Granig
Michael Prokop
szcom
Razvan Crainea
hdikme
Victor Seva
John Burke
Sebastian Kemper