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
Distinguish between unconfirming the learned peer address and
retriggering the kernel stream. In particular we don't want to unconfirm
the sinks every time we confirmed our own peer, as that starts an
unconfirm/reconfirm loop.
Change-Id: I1f172385aefeacbc4585729bce25fbc68f04c2bd
While doing the A/B reassociation during an offer/answer exchange, we
don't (necessarily) want to remove all existing subscriptions. Instead
we cant to unsubscribe all subscribers so we don't do media forking, but
leaving existing subscriptions alone to make early media reception
possible. This mirros the old behaviour.
Change-Id: Ib9e6671ca2d23d1eb4509d7cf939015c816cc622
Multiple untagged monologues can exist at the same time which would lead
to a broken bencode dictionary. Instead use a pseudo label to
distinguish them.
Change-Id: I0f41c42df8ec17c1c4fb5cc6451ea039612e505f
We may have multiple subscribers, some of which may be dead/unused. We
don't care if we have these since we don't forward to them anyway.
possibly relevant for #1337
Change-Id: I3cded5080aa2005e9dd615cccf60bd4cba5feb7d
Set NO_KERNEL_SUPPORT when we don't actually kernelise the stream, and
use that flag when trying to pull stream stats.
probably closes#1337
Change-Id: I46af55e353d87c5afdda3c106d1f3470273105bf
The advertised address might be empty (trickle ICE) so use the FILLED
flag instead to see if the sink is eligible.
Change-Id: I114bd7400ccfcc3ecbc871bdcc5aee4e7d699816
Make sure janus_session lock is obtained first and websocket_conn lock
second, in order to prevent a possible deadlock.
Change-Id: I3db1d5cea0c0295cc10c71edd20c86ce054f520b
Warned-by: Coverity
This isn't really necessary as at this point the janus_session object is
private to the thread, but we add the locking anyway to silence the
warning.
Warned-by: Coverity
Change-Id: I0b192f2f5827ad917cf5110ce486fc6cd49e1a71
If a keyspace notification SET is received and the call already exists
as a foreign call, the call is first destroyed before being re-restored.
The call destruction involves a DEL from Redis on the "hosted DB"
number, which points to the foreign DB. This makes it impossible to then
restore the call because it's just been deleted.
closes#1308closes#1334
Change-Id: Ie895b021441b2d299f8ebb5bde1824b01e12633c
Also add a safeguard against filling the remote peer address with an
address from the wrong family
closes#1305
Change-Id: Iac18212b4d526a2f7d49a06ddcd724aa89b06060
The contents of the ->next element cannot be accessed completely lock
free as they're zeroed out during call removal. Instead grab a reference
to the linked next call before releasing the lock, and also lock the
next element before moving on. This requires a more granular locking as
not to interfere with call removal: One lock to protect the contained
call and the ->next, and another to protect the ->prev
Change-Id: I5474ea3f88e3276f93ba62a952b3be13c0c182e9
commit b0c722da69ad088a2eddced12b37c0546a514890
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 15:35:51 2021 +0200
changed flag bit length
in call_interfaces.h changed bit length of reuse_codec from 0 to 1
commit 0313a747532d5987f25fa9edb202aa460bf98dd1
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 15:29:20 2021 +0200
inversed reuse_codec logic
in test-transcode.c and call.c, reuse_codec = 0 (default) will now result in using codec_store_populate instead of codec_store_populate_reuse
commit b876bd686bd30df21a5962aca16fc1c85574f554
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 15:18:19 2021 +0200
adding option to minimalize changes in the codec_store_population
added function codec_store_populate_reuse in codec.c which replaces codec_store_populate but makes fewer changes to the GLists with the old and new codecs
added flag to enable this feature (disabled by default)
commit 6fd0b701c9589b2fae00300801e02a9b5cc397ab
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 14:44:42 2021 +0200
Added Option to minimize change in the codecs
In codec.c added function to populate codec store with the fewest changes between the old and new GList which contains the codecs.
Added new testroutine in test-transcode.c line 1500
Added flag to call_interfaces.h to optionally enable this feature
Change-Id: If58d9a07d114b05dfb75553a87eb4372ae949fbb
commit 3bf554a8fbae7e948343699f40d935693618b764
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Fri Jul 23 13:58:02 2021 +0200
changing codec-exchange behaviour
in codec.c line 3288 function codec_store_populate now doesnt empty dst and copy new codec from src to it, instead codecs from src will be appended to dst and codec from dst, which are not
being contained by src are being removed
Change-Id: Id6b7ee65595f9cc5c71ef557c7bac5ee38f97cbe
This restores backwards compatibility with existing code parsing the
output, e.g. Kamailio to report MOS stats.
Change-Id: Ibafb2a7a3cf118166ffe1cca9a62a06b63252944
This fixes a race condition: Peer sends updated SDP with new address,
but an older RTP packet from the old address is received afterwards.
Thsi triggers learning of this old address is the "correct" endpoint.
Afterwards the peer stops sending RTP until a packet to the new endpoint
is received there, which never happens because the new endpoint has been
discarded in favour of the "learned" old one.
closes#817
Change-Id: I508f465a669f03e35ddcc6e770d5e7859e57569f
commit d15fd4a547
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Tue Jul 6 14:07:42 2021 +0200
Also free the character data.
commit 8869187215
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Mon Jul 5 16:11:32 2021 +0200
Make documentation and parameters consistant.
commit 4b15aea2ee
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Thu Jun 10 15:34:00 2021 +0200
Update documentation.
commit 6ec1b3035d
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Thu Jun 10 15:33:12 2021 +0200
Do not use metadata for setting the recording file; use output_destination instead.
commit f65a76e8a3
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Wed Jun 9 15:56:02 2021 +0200
Only append file extension if skip_filename_extension is not set.
commit 92e9d7c679
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Wed Jun 9 14:12:48 2021 +0200
Rename the option to better match its usage.
commit 11128bff49
Author: Damir Nedžibović <damir.nedzibovic@enreach.com>
Date: Wed Jun 9 13:19:10 2021 +0200
Implement support for seting an output file and folder per recording.
Change-Id: I1579d62467eaf06a7aa1ac11e59dbb374f150deb
Move RTCP processing down into the egress section and run the processing
function once for each output, because the output media must be known
when processing RTCP RRs during transcoding.
closes#1298
Change-Id: I1797bef336e27a7064b9f42ab8c25f0aade02e47
Perform reverse SSRC mapping even if output media is not known, and use
the appropriate media side when looking up SSRCs for received RTCP
timestamps.
closes#1298
Change-Id: Ifa5a982163bf7b0510ffc2a92ae25995d1adb888
Complete overhaul of the codec handling code:
*) obsolete flags `asymmetric codecs`, `symmetric codecs`, `reorder
codecs`
*) support proper codec offer/answer
*) split codec manipulation (strip/offer/accept/etc) into separate
functions for clarity and better code maintenance
*) fully update codec handlers in both directions after an answer
*) explicit allocation and handling of codecs and payload types in a
codec_store object
*) improve codec matchup logic during answer
*) more explicit handling of supplemental codecs (CN/DTMF)
*) remove now obsolete hacks for handling certain use cases
Change-Id: I996705ba8fe339524c2f70e6bb0fd854f9a1f4fb
This solves problems when the same SSRC is looped through the same call
multiple times in different mono/dialogues, with different parameters.
Change-Id: I1d033cb1f012574d82b5bcbfffe11eb5f983cfd8
Transcoding should not be decided based on the name of the codec alone,
but primarily on the payload type. First the PT needs to be compared,
then the codec type must be confirmed.
closes#1289
Change-Id: I1a8bffc6d521443aba14d9b4cf1ad4d1e21f1226
The JSON context is already in an object, therefore skip the surrounding
{} enclosure.
Also don't run the global MQTT timer when MQTT is disabled.
closes#1290
Change-Id: I63c622bd339545e625ee18def33f21de2533f949
Richard Fuchs
hdikme
Stefan Mititelu
Victor Seva
John Burke
Sebastian Kemper
Ricardas Stoma
Daniel Hauptmann
Damir Nedžibović