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
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
Older versions of glib handle the trailing padding in base64
differently, resulting in the final character of the encoded string
coming out different.
Change-Id: I2bc1057de15f5e0bddb3de4df6cef7c4fcf7ebcc
We have to cover more cases with tests, such as:
- SDES re-ordered crypto suites, but one suite from the 'SDES-order:'
flag was not in the offer
- SDES re-ordered crypto suites, but one suite from the 'SDES-order:'
flag was not in the offer and the recipient selected it
Change-Id: I1051c01d45c4b494f768692f85d1e0c41a0ea2d2
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
This type is used as const everywhere except internally, so make it part
of the typedef for brevity.
Change-Id: Ic4afe037b392239a991d5380c6708903011da29e
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
Support fake bind() on non-INET sockets. This fixes the tests when
building against libwebsockets19 4.3.2-1.
Change-Id: I91e5271e8b0cd5a0fa10317bae059615b234926c
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
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
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
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
related to #1497#1549
> ======================================================================
> ERROR: setUpClass (__main__.TestWSJanus)
> ----------------------------------------------------------------------
> Traceback (most recent call last):
> File "/code/t/auto-daemon-tests-websocket.py", line 114, in setUpClass
> eventloop.run_until_complete(get_ws(cls, "janus-protocol"))
> File "/usr/lib/python3.10/asyncio/base_events.py", line 646, in run_until_complete
> return future.result()
> File "/code/t/auto-daemon-tests-websocket.py", line 23, in get_ws
> cls._ws = await connect(
> File "/usr/lib/python3/dist-packages/websockets/legacy/client.py", line 622, in __await_impl__
> transport, protocol = await self._create_connection()
> File "/usr/lib/python3.10/asyncio/base_events.py", line 1089, in create_connection
> transport, protocol = await self._create_connection_transport(
> File "/usr/lib/python3.10/asyncio/base_events.py", line 1107, in _create_connection_transport
> protocol = protocol_factory()
> File "/usr/lib/python3/dist-packages/websockets/legacy/client.py", line 160, in __init__
> super().__init__(**kwargs)
> File "/usr/lib/python3/dist-packages/websockets/legacy/protocol.py", line 154, in __init__
> self._drain_lock = asyncio.Lock(
> File "/usr/lib/python3.10/asyncio/locks.py", line 78, in __init__
> super().__init__(loop=loop)
> File "/usr/lib/python3.10/asyncio/mixins.py", line 17, in __init__
> raise TypeError(
> TypeError: As of 3.10, the *loop* parameter was removed from Lock() since it is no longer necessary
>
> ----------------------------------------------------------------------
Change-Id: I3178c54ed7eb40b9cc06769c1f1e237e0d58f966
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
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
The order between receiving the STUN success and the triggered check is
not guaranteed, therefore resolve possible race condition by expecting
the two packets in either order.
Change-Id: Ibef9907cd4116bc5f3b7d17d936007c8efcabd3b
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
* Use an explicitly created global event loop
* Await websocket connection closure
Relevant to #1497
Change-Id: I600189f5383ca7e5da8b45460508c1ddcddede0b
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
This eliminates a spurious false warning log message for rejected
streams that use a dummy payload type
Change-Id: Id628cafb8d7c4ea576cd01ff35f5dd9cd2151280
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
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
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
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
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
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
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
These tests are timing sensitive which makes them unstable under certain
conditions. Remove them from the automated build system.
closes#1309
Change-Id: I432445bce337bbf4d4b80417e532a910b516b8ee
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
We should eventually try to reduce the amount of -Wno-* options, but for
now this is a net improvement.
Change-Id: I3bd03679acbc157c0d1b3c257a542e2eec0e5ee9
This makes the type in line with string(3) functions and eliminates some
compiler warnings.
Also update the related bencode data type.
Change-Id: I7ef4024f4b5a0f737b3dbe03bcd078032395bce6
Since dh_auto_test doesn't execute the test suites through make
directly, but instead runs `make -s -n` and then executes the output,
the integrated build tests fail since the sub-make doesn't return an
error as it should when attempting to build with the wrong .h
alternative, resulting in always the first .h alternative being used.
Fix this by using a wrapper script instead.
Also adjust some other related minor build details.
closes#1202
Change-Id: I4b6436295c6b39117bd06df53aa5afc7118ad6a1
Whether a bit-field is signed or unsigned is implementation specific, so
we should be explicit about this.
Change-Id: I744df3d24bc08e95fa816ba4135f19cd3a5dcb17
Warned-by: lgtm
While these usages are not supposed to be security sensitive, it's still
best to avoid the usage altogether so that we do not need to think about
it. Add a taint macro so that this does not regress in the future.
Change-Id: Ic75861ed1b7ce9dfca4f897de8be2408204ce3cb
Warned-by: coverity
Rename test suites from *-test.c to test-*.c as the former is
conflicting with the file name pattern of the build tests, which are
listed in .gitignore.
Change-Id: I8779fac492510c0996bdcb2c6250c07536fe3b63
Avoid using an explicit test for the trickle ICE endpoint address and
instead use the TRICKLE_ICE flag.
Change-Id: I821ced7e77c798004c9d88ec072848b7f6e24a85
close PR #1133
Squashed commit of:
commit b5009eff0b
Author: SPChan <shihping.chan@gmail.com>
Date: Fri Dec 11 11:40:17 2020 +0800
crypto.c: handle 12 bytes master salt length of AEAD
AEAD RFC7714 specifies 12 bytes master salt length
(section 12) but the KDF expects 14 bytes.
The erratum to RFC7714 specifies to pad on the right
by 16 bits to be compliant with the KDF.
https://www.rfc-editor.org/errata_search.php?rfc=7714
commit 30ba71f10c
Author: SPChan <shihping.chan@gmail.com>
Date: Fri Dec 11 09:31:00 2020 +0800
crypto.c: AEAD make clear which operations are for AAD
commit 3e4d87681d
Author: SPChan <shihping.chan@gmail.com>
Date: Thu Dec 10 19:38:51 2020 +0800
Implement RFC7714 ciphers: AEAD AES
Initial userspace only implementation
Change-Id: I8624b255e4fd83a5ca418655ef117c48e56e2379
If a supplemental codec was removed by the codec tracker, we must reset
any codec handlers on the opposite side that were using this output
codec.
Change-Id: I7fb9d0ec7a061682fe3b28d9ead14ac263aff7c3
Link the attribute value to each RTP payload type and only output it if
PT is still present
Also merge PT attribute printing into a single loop
closes#1128
Change-Id: I494498ca806b795045e9e5bbb82174fa4aef8b2b
Additionally allow `except` and `offer` (and `transcode`) to revert
masked codecs.
Also adds `mask=full` and `strip=full`
closes#1054
Change-Id: I93a2891f2a0b9d324f6804e38f992be73c46211a
An endpoint switching protocols is normally passed through to the peer,
but in an answer that is usually not desirable. Change the default
behaviour to stick to the original protocol that was offered even if the
answerer changes protocols.
Change-Id: Ib288549f4b1c9ab57a6333c6b7dd511537af96f7
Initially created branches don't have a tag associated with them, so
they don't appear in `other_tags`. We need to keep track of which
branches were created from a single from-tag through their via-branch
values so we know when all branches have been deleted.
closes#1037
Change-Id: I4c6ce602b99a9104de98098cd06ca399eb28e1b2
1) In an SRTP re-invite offer, fixes not a full set of crypto suites
being present
2) In a re-invite offer that switches from RTP to SRTP, fixes SRTP not
being initialised at all
Change-Id: I911442d2cba17ecf6af482cfe922d4e9db2eda8d
Richard Fuchs
Donat Zenichev
Victor Seva
Razvan Crainea
Stefan Mititelu
hdikme
Daniel Hauptmann
Guillem Jover
Damir Nedžibović
SPChan