low/high layer compatibility are used for capability checking between
caller and called entitiy. The transcoding is performed by libosmogsm.
Related: OS#6152
Depends: libosmocore.git Ia6a2159ecf810a02f85b558026edf20b934567de
Change-Id: I760980a7e17e2fa81615adc69ef85797eb0c07f1
This is the last missing piece that allows osmo-msc to make good TFO
codecs choices.
Since the codec_filter, osmo-msc properly gathers codec options and
limitations. But the MO call leg still assigns a voice channel before
getting a response from the MT call leg, and is then stuck with that.
Add the capability to adjust the MO call leg's codec in case the MT side
needs a different codec for TFO.
This is only relevant for 2G; on 3G we always have AMR/IuUP.
For inter-MSC handover, keep the behavior unchanged: offer only the
currently assigned codec to the remote side. Codec-changing HO should be
equally trivial to implement, but that is for another day.
msc_vlr_test_call's codec tests are adjusted to test the new feature in
Ib933554f826c1b4347dfa3f6c4f6fe086be8b133. For now, avoid change in
these tests by validating the first codec in SDP lists only.
Related: OS#6258
Related: osmo-ttcn3-hacks I402ed0523a2a87b83f29c5577b2c828102005d53
Change-Id: I8760feaa8598047369ef8c3ab2673013bac8ac8a
To parse and handle SDP included in incoming MNCC, use rx_mncc_sdp()
everywhere. So now rx_mncc_sdp() is the single implementation for
parsing the SDP string and taking action for codecs if needed.
One current dup of this code has a fall-back to use legacy bearer cap --
absorb that into rx_mncc_sdp(), so that we now also do that fall-back
for all of the incoming MNCC that contains bcap.
This is a cosmetic preparation for implementing MO Re-Assignment to
match MT's codec limitations.
Change-Id: I94ae11654e1f88fbd64361b639a4c583836dc13e
Bearer capability 3k1_AUDIO and FAX_G3 are only important
for the interworking function, the MSC should handle
these calls the same as CSD calls with unrestricted digital
bearer capability.
Change-Id: I198aa867a8f236b8ddd05d3b2356f64b876fd4c1
The intent of the guard timer is to clear hung or stuck states
during call setup or teardown. However, there are some MNCC
messages that will be exchanged between OsmoMSC (passing CC
messages to and from the MS) and the external MNCC agent during
the active call state, not related to setup or teardown: DTMF
start and stop, plus call hold and retrieve operations for call
waiting. Unpatched OsmoMSC restarts the guard timer on every
received MNCC message, even those that pass through to CC without
affecting any state, and the result is breakage for users.
Consider the case of an IVR where you have to press some DTMF keys
before you can be transferred to a human operator. You press the
needed keys, get the human operator, and start talking. Then
3 minutes into your conversion (default guard timer duration)
your call unceremoniously disconnects without any warning.
Fix: look at the MNCC message type, and skip the call to start
the guard timer for known-benign MNCC messages.
Change-Id: Ibe2dd53f8e9e06d175b64df67d2a2e3e2d4155aa
Fail if MNCC tries to switch the Information Transfer Capability from
CSD to speech, so it is obvious that something is wrong here. I ran into
this while writing a test.
Related: OS#4394
Change-Id: Ibb76d08cad1ac3bc3320391c89766150a2e605c3
A transaction can be identified by the callref and the type. Because
transactions with different types may share the same callref value,
it is required to include the type in the trans_find_by_callref()
parameters.
E.g. a voice group call may have the same callref as a voice broadcast
call, but they are different calls. They also may not be confused with
other transaction types having eventually equal callref value, like
GSM 04.08 calls, SMS or supplementary services transactions.
By adding the transaction type to trans_find_by_callref(), we
essentially now use the (type, callref) tuple as unique ID for
transactions, instead of just callref.
Change-Id: Ic0b82033a1aa3c3508ad610c690a5f29073006c1
Related: OS#4854, OS#3294
The MGCP protocol features the 'C' (call-id) to identify which
connections belong to the same call. They may be used by MGW for
accounting or management procedures.
So far we sent the MNCC callref as call-id. Instead, add a separate
unique call_id number space. Assign a unique call_id to each
transaction.
Change-Id: I36c5f159fa0b54fb576ff8bd279928b895554793
Related: OS#4854
Use the MNCC bearer capabilities in CC setup for CSD, if available.
Note that in the MNCC_F_BEARER_CAP code path sdp_audio_codecs_set_csd()
also gets called by trans_cc_set_remote_from_bc().
Related: OS#4394
Change-Id: I56e49ebc41696912a81b8f4f63fbc36d0b605e9e
Prepare to use trans->bearer_cap.transfer in trans_cc_filter_run() to
differentiate between speech and data (CSD).
Related: OS#4394
Change-Id: Id0476a4882bcb27413d033f2de2c5288954f0b95
Move remote out of codecs, as it will be used by CSD code as well.
Otherwise we would need to store it twice (in cc.codecs.remote and
cc.csd.remote).
Related: OS#4394
Change-Id: I5d2e078db3b3437cb6feae40d8955912d7a297e4
Remove the comment as trans->bearer_cap will be used in CSD code to
differentiate between speech and data.
Related: OS#4394
Change-Id: I0539632f464bc44945599bec52dc2a4df2f0115f
Remove the misleading "We must not pass bearer_cap to
codec_filter_init()" part of the comment. The function doesn't accept a
bearer_cap parameter, it cannot be passed to the function:
void codec_filter_init(struct codec_filter *codec_filter)
{
*codec_filter = (struct codec_filter){};
}
Related: OS#4394
Change-Id: I87a1e371e108d8da514b30f1726aad0f85ea4111
In all the places where codec_filter_ functions get called, for CSD we
will need to filter the bearer services. Add a new
transaction_cc.c file for functions that either combine the
codec_filter_ function with logic for CSD and voice calls or just call
the existing codec_filter function and a new csd_filter function.
Start with moving codec_filter_set_ms_from_bc to this new file, it will
be extended with a case for CSD in a future patch.
Related: OS#4394
Change-Id: If225f2a299ce6bc9ae35a17d6f591d889f49155e
In order to send the MSC's RTP endpoint IP address+port in the initial
SDP, move the MGCP CRCX up to an earlier point in the sequence of
establishing a voice call.
Update the voice call sequence chart to show the effects.
Though the semantic change is rather simple, the patch is rather huge --
things have to happen in a different order, and async waits have to
happen at different times.
The new codec filter helps to carry codec resolution information across
the newly arranged code paths.
Related: SYS#5066
Change-Id: Ie433db1ba0c46d4b97538a969233c155cefac21c
Transmit and receive full SDP information via MNCC, to accurately pass
codecs choices between the call legs.
In msc_vlr_test_call.c test_call_mt(), show that when receiving MNCC,
the codec information in SDP overrules the Bearer Cap codec information
-- we expect to still receive inaccurate Bearer Cap from e.g.
osmo-sip-connector, because we have chosen to add SDP to MNCC instead of
trying to fix the codecs represented in Bearer Cap.
For internal MNCC, the MT call leg now knows which codec the MO has
chosen and assigned.
For external MNCC, osmo-sip-connector receives SDP about our codecs
choices and sends it in SIP messages, and we also receive the full SDP
information from the remote SIP leg.
Update the SDP in codec_filter every time it is received, to always have
the latest SDP information from the remote leg.
CC MNCC
| ---ALERTING--> | add local side SDP to MNCC msg
| <--ALERTING--- | store remote side SDP
| <--SETUP-RESP- | store remote side SDP
| --SETUP-CNF--> | add local side SDP to MNCC msg
| -RTP-CREATE--> | use codec_filter, add local side SDP to MNCC msg
| <-RTP-CONNECT- | store remote side SDP
There still is one problem: when initiating MNCC, we do not yet know the
RTP address and port to be used for the CN side, because the CN CRCX
happens later. So far we send 0.0.0.0:0 as RTP endpoint in the SDP,
until the CN CRCX is done. A subsequent patch moves CN CRCX to an
earlier time, adding proper RTP information right from the start.
Related: SYS#5066
Change-Id: Ie0668c0e079ec69da1532b52d00621efe114fc2c
So far, patches have set up rtp_stream to allow setting multiple codecs,
and collected the codecs information into the codecs filter struct.
Now actually use the codecs filter result to choose a codec.
Setting up the call leg FSMs and codecs still looks rather confusing in
this patch, because this is an incremental step in a larger series. The
upcoming patch 'do CN CRCX first' clarifies this substantially.
The resulting codecs behavior is tested in upcoming patch
I879ec61f523ad4ffc69a0b02810591f7c0261ff9. (The test ideally should have
come before this patch, but my time to rework this branch is up.)
With the codecs filter in place, we are ready for sending and receiving
full SDP via MNCC, see upcoming Ie0668c0e079ec69da1532b52d00621efe114fc2c
and Ie433db1ba0c46d4b97538a969233c155cefac21c
Related: SYS#5066
Change-Id: I66e7c8c5e401f4f3a7d3d42b9525b2c6e99691d9
So far, we just forwarded the Bearer Capabilities received in MNCC from
the remote MO call leg, and omitted Bearer Cap if the remote call leg
did not provide any.
Instead, always include Bearer Cap, and compose it from the codecs
filter result. Hence the Bearer Cap is now an intersection of MS, BSS
and remote call leg, instead of just the remote call leg.
Related: SYS#5066
Change-Id: I9586221ef56352b7ce4b2604ae0dc04554145a78
Allow configuring MGW conns with multiple codecs. The new codecs filter
can have multiple results, and MGCP can configure multiple codecs. Get
rid of this bottleneck, that so far limits to a single codec to MGW.
On Assignment Complete, set codec_filter.assignment to the assigned
codec, and use that to set the resulting codec (possibly multiple codecs
in the future) to create the CN side MGW endpoint.
Related: SYS#5066
Change-Id: If9c67b298b30f893ec661f84c9fc622ad01b5ee5
Collect either the SDP or the Bearer Capabilites in the incoming
MNCC in the new codecs filter.
So far just collect the info and do not change the behavior, using the
filter result will follow in a subsequent patch.
Related: SYS#5066
Change-Id: I84d9bbca3e4061da622b1b2fc0bde8868e7e3521
For MT call, initialize the codecs filter and apply the
Codec List (BSS Supported) from Compl L3.
Related: SYS#5066
Change-Id: I530409a64d11da48518a3dc60aa3a4e47c384663
Omit "in state FOO", because LOG_TRANS() already logs the state.
Most MNCC "rx" logging was duplicated. Log "rx" only once.
If there is RTP information passed with the MNCC message, log it:
- if there is SDP, log the SDP information.
- if there is no SDP, log the legacy MNCC RTP fields, if any.
One motivation to do this is to get RTP information in ladder diagrams
generated by msc_log_to_ladder.py without the need to add udtrace MNCC
logging to osmo-msc; and also to get RTP info for internal MNCC, where
udtrace doesn't apply, because no unix domain socket is involved in
internal MNCC operation.
Change-Id: I4b916cb482ed441b508c6295de211a21c49cd5c1
The timer "mgw X2" (RTP stream establishing timeout)
is set by default to 30 seconds.
When an MT call is ringing and remains unanswered, it
is this timer that will expire, and the call is terminated.
Up to now this results in a CC_CAUSE of Resource Unavailable
and if osmo-sip-connector is in use, the SIP agent will
get 503 Service Unavailable.
While "resource unavailable" may be technically correct, in
that the MGW did not return an rtp stream in time, returning
"No User Responding" (resulting in SIP 480) is probably a
more accurate description of what actually happened,
allowing the switch to inform the caller.
Change-Id: I4a9cfc388ec9ecb743d154a114a6db638eac4701
Using *unpacked* 'struct osmo_gcr_parsed' in the MNCC PDUs makes
the protocol even more complicated than it currently is, and
moreover complicates implementing MNCCv8 in the ttcn3-sip-test.
Replace 'struct osmo_gcr_parsed' in 'struct gsm_mncc' with a
fixed-length buffer, which is supposed to hold the Global Call
Reference encoded as per 3GPP TS 29.205.
Indicate presence of GCR using the MNCC_F_GCR flag.
Change-Id: I259b6d7e4cbe26159b9b496356fc7c1c27d54521
Fixes: I705c860e51637b4537cad65a330ecbaaca96dd5b
Related: OS#5164, OS#5282
This commit is largely based on work by
Max <msuraev@sysmocom.de>
Adds LCLS parameters for A-interface transactions
This commit also adds a vty option to facilitate globally
disabling LCLS for all calls on this MSC.
Add a global call reference (GCR) to MNCC and therefore
bump the MNCC version to version 8. (This commit has to be
merged at the same time as the corresponing commit in the
osmo-sip-connector for mncc-external use.)
Depends: osmo-sip-connector Id40d7e0fed9356f801b3627c118150055e7232b1
Change-Id: I705c860e51637b4537cad65a330ecbaaca96dd5b
Calling gsm48_cc_tx_release() before mncc_release_ind() has a side
effect: the former may change CC state to GSM_CSTATE_RELEASE_REQ.
This makes the later send MNCC_REL_CNF instead of MNCC_REL_IND, so
if one of the call leg disconnects due to RF failure, the other one
will not be terminated correctly.
Makes both TC_{mo,mt}_call_clear_request TTCN-3 test cases pass.
Change-Id: I3ad4a99757878de3796027325627c87d9a4e93f1
Related: Id16969fe0de04445d1320a96d35cf1d48cc8cf09
Related: SYS#5340
Do not free the CC transaction when an MT subscriber is already being Paged.
Instead, invoke another paging request, which paging.c will correctly add to
the list of pending paging response callbacks to run.
A ttcn3 test is linked in the related patch (s.b.).
Related: OS#4240
Related: Ieeae6322d4e80893ea3408c6b74bf8e32bea8e46
Change-Id: Idd4537b5f4817d17e5c87d9a93775a32aee0e7be
Use of this flag was dropped when adding inter-BSC and inter-MSC Handover
support, I forgot to remove it.
Change-Id: I5ec78e30eb36fbe78a3f7c46bfa44af5a4eb7bf2
If an incoming MNCC_SETUP_REQ ends up in Paging (as usually it does), the early
return so far skipped logging of that MNCC message. Add this logging.
Change-Id: I1495dd562a06cf6c1e9453a1fe111bdf8f4be081
Andreas Eversberg
Neels Hofmeyr
Manawyrm
Oliver Smith
Keith Whyte
Vadim Yanitskiy
Pau Espin
Martin Hauke