When a subscriber first attaches by TMSI only, and later tells the IMSI
via ID Response, it may turn out that this IMSI already exists in the
VLR database. If this happens, the TMSI that the subscriber issued was
not known in the existing VLR entry, indicating that the subscriber has
in the meantime camped on a different core. Which means we can assume
that there cannot be any active connections, and the old subscriber can
be discarded, for the benefit of the new one.
(We could also discard the new one, but it is more complex to reparent
the ongoing FSMs for Compl L3 than to copy some dormant VLR state.)
In vlr_subscr_set_imsi(), check for an existing IMSI entry in the VLR.
If such exists, copy any pending Paging and auth tuple state to the new
subscriber, and discard the old one from the VLR.
In order to safely discard a vlr subscriber by force, add a new vlr_ops
function: subscr_inval(), to tell the MSC that a vlr_subscr is no longer
valid.
Upcoming patch I583682d1a35a70b008d7bb2d89ba7c3109a60b21 better clears
TMSI state from the VLR, making it more likely to hit the evil twin
situation this patch fixes; hence this is, sort of, preparation.
Related: SYS#6860 OS#4721
Change-Id: Ifdabe0b65bffafbf7b8e5cc10e2d225d1ed1cecd
Better match the pattern of sdp_audio_codecs_* instead of having
foreach_ in the front. Prepare for prepending osmo_ some day, because I
plan to move the SDP API to a separate library.
Change-Id: Ia96190e0bdb513886663be1c8c12be3b403b71c9
When we get the codec filter result logged, it is most interesting to
know the caller. So wrap a file-line macro around trans_cc_filter_run().
Change-Id: I243404487c1871e921b08098086ef2fc78a5561d
low/high layer compatibility are used for capability checking between
caller and called entitiy.
The information is added to the end of struct gsm_mncc increases, so
that the version number needs not to be incremented.
Related: OS#6152
Change-Id: I15f5afcf069ee6c1c4641108ceacc837bee311b5
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
Used by I8760feaa8598047369ef8c3ab2673013bac8ac8a to add just a single
codec to a speech codec list, instead of a list.
Change-Id: I6ac23c54bc26939e048ff2df06eb987421cfb1c5
When trying to modify the value of an SGs counter (eg. ns11), then the
setting is never stored. The reason for this is that OsmoMSC uses the
wrong string table to compare the user input.
Related: OS#6008
Change-Id: I0358c1ec0026c37fda6db1f3af3145393df25cfd
For MO-forwardSM and MT-forwardSM request messages, OsmoHLR applies
routing based on the SMSC address for MO or based on the IMSI for MT.
However, reply messages following these requests are routed passively
based on the destination_name IE. This passive message routing path
requires the source_name IE to be set as well - implement this
source_name setting.
Related: OS#6135
Change-Id: I0b7f4760bdce8a38d43d3860086c6dfb7b390701
When OsmoMSC is used with OsmoHLR rather than a GSUP-to-MAP gateway,
MT-forwardSM.req GSUP messages delivering MT SMS will be coming from
a separate SMSC relayed via OsmoHLR, rather than from OsmoHLR itself.
When we reply to these messages, in order for these replies to reach
the MT-sending SMSC via OsmoHLR, we need to save source_name from
the request and regurgitate it into destination_name in our response
messages. Implement this logic.
Related: OS#6135
Change-Id: I436e333035b8f6e27f86a49fe293ea48ea07a013
Switching ASCI support is controled via VTY. This added in a later
patch. (Chg-Id: I5bd034a62fc8b483f550d29103c2f7587198f590)
Change-Id: Id68deb69f7395f0f8f50b3820e9d51052a34f753
Related: OS#4854
A voice group/broadcast call has no SCCP connection that is related 1:1
to a calling or called subscriber. Instead there are multiple connections
between MSC and BSS. Some of them control the uplink for each BSS and
some of them assign the channels for each BTS.
SCCP connections are maintained by the VGCS call control. Message from the
RAN are directly forwarded to the VGCS call control.
Change-Id: Ie4a2f19ba75140a6f2de02b709597239c01f02a2
Related: OS#4854
The (optional) call reference is required to assign a calling subscriber
to a voice group/bcast channel. The BSC can then determine to which
existing VGCS/VBS channel the MS is assigned to.
This IE is part of the GSM standard TS 48.008 (see §3.2.1.1)
Change-Id: I7955c6e0eebc930f85f360dda46be17cbd39e181
Related: OS#4854
This is a built-in data structure to store and handle voice group calls.
The GCR will be used by VGCS/VBS call control.
(Chg-Id: I9947403fde8212b66758104443c60aaacc8b1e7b)
The GCR will be used by VTY code.
(Chg-Id: I5bd034a62fc8b483f550d29103c2f7587198f590)
Change-Id: Ia74a4a865f943c5fb388cd28f9406005c92e663e
Related: OS#4854
- TRANS_GCC is used for the voice group call.
- TRANS_BCC for the voice broadcast call.
This also includes the use counters for transaction and CM service
request usage:
- MSC_A_USE_GCC
- MSC_A_USE_BCC
- MSC_A_USE_CM_SERVICE_BCC
- MSC_A_USE_CM_SERVICE_GCC
Change-Id: Iddd11f813582ac2ac2bdee91cc3a525986deb514
Related: OS#4854
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
Allow the caller of rtp_stream_alloc() to define what events will be
dispatched to the parent FSM. This allows other state machines to use
rtp_stream. It is required for using RTP stream process with VGCS FSM.
Drop the unused parent_call_leg member.
Change-Id: I0991927b6d00da08dfd455980645e68281a73a9e
Related: OS#4854
So far rtp_stream_commit() triggers an MGCP MDCX message only when
codecs or the RTP address changed.
Do the same for mode changes. ('sendrecv', 'recvonly', 'sendonly',...)
Change-Id: I7a5637d0a7f1df13133e522fc78ba75eeeb2873e
Related: OS#4854
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
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
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
Do not convert to enum mgcp_codecs, but directly pass the
gsm0808_speech_codec IE from the A interface to codecs handling.
For Iu:
- RAN side: use ran_infra.force_mgw_codecs_to_ran to keep the MGW
endpoint towards RAN on IUFP.
- CN side: introduce flag ran_msg.assignment_complete.codec_with_iuup,
so to decide whether to forward IUFP towards CN, we don't need to test
the RAN type, but use the flag from the ran_msg implementation.
In msc_vlr_tests, use the SDP codec string instead of enum
mgcp_codecs.
So far limit to intra-MSC related messaging, adjusting inter-MSC
handover follows in a separate patch.
Change-Id: Ia666cb697fbd140d7239089628faed93860ce671
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
Indicate in the ran_infra data structure whether a RAN needs specific
codecs to be set up on the RAN facing MGW endpoint.
This allows setting forced RAN codecs as first-class citizen in the
ran_infra data structure, instead of special cases in the code (for IuUP
on IuCS).
Will be used in subsequent commit
I37f65c36af2679ecba1040a11a9aa0eb9481d817, submitted separately for
easier readability.
Change-Id: I37f65c36af2679ecba1040a11a9aa0eb9481d817
The initial Compl L3 happens long before we establish a CC transaction.
Remember the Codec List (BSS Supported), so that we can feed the new
codecs filter with it. Subsequent patches implement feeding the filter.
Related: SYS#5066
Change-Id: I7cdc348218433141a43d2e42750af02591688240
Add the central codecs_filter for Call Control. The new member is not
used in this patch yet, subsequent patches will start to populate the
various stages of this codec filter, one by one.
Related: SYS#5066
Change-Id: Ib3fdeff8d1e1ea0760168d63ee6e1b1fb993aa5f
Add the infrastructure to store and filter all codec limitiations from
the different stages: MS, BSS, CN and remote call leg. Upcoming patches
will properly collect these and find an optimal codec.
No functional change, yet.
Related: SYS#5066
Change-Id: I4d90f7ca62f2307a7b93dd164aeecbf4bd98ff0a
Converting between different codec representations is confusing. This
codec mapping provides a consolidated overview of all our codec
representations, and how they match up.
In particular, it adds the SDP codec representation repertoire,
preparing the use of full SDP on the MNCC interface.
Related: SYS#5066
Change-Id: Iaa307be6a8487aa8d4ba7cd59d5c5ef04818a744
Neels Hofmeyr
Andreas Eversberg
Philipp Maier
Oliver Smith
Pau Espin
Vadim Yanitskiy