ptmap[].codec is already set correctly, so we can simply drop use of
codecs[] now.
Related: osmo-mgw I798e02c6663376d3d52f4a74fc4b32411ce95bed
Change-Id: I701eb97ddafa871cfbc12fdbe8a1fe14aa2a4db1
Since recent libosmo-mgcp-client update (see Depends), we can now pass
fmtp for each individual codec entry to be sent in the SDP. Use that new
feature to pass any fmtp from rtp_stream.c to the MGW via MGCP.
In effect, this adds the 'octet-align=1', defined in codec_mapping.c as
the default for AMR, which was missing from the SDP/MGCP that osmo-msc
sends out.
A 2G to 2G call worked out OK without it, because omitting the
'octet-align' setting from MGCP disables all octet-align payload checks
in osmo-mgw. Both 2G call legs use octet-align=1, and things work out.
Running a call from 3G to 2G before this patch would result in osmo-mgw
dropping the RTP with logging like:
DRTP NOTICE (rtpbridge/1@bsc0 I:B38CF524) rx_rtp(44 bytes): Expected RTP AMR octet-aligned=1 but got octet-aligned=0. check the config of your call-agent! (mgcp_network.c:1527)
...because osmo-bsc's MGW was instructed to expect octet-align=1, while
osmo-msc's MGW emits octet-align=0. This happens because converting the
3G IuUP to 2G AMR heeds the RFC 6871 specified default of octet-align=0.
So, after this patch, we accurately send "octet-align=1", and make the
IuUP-to-AMR conversion emit octet-align=1 to match 2G.
Side note: 3GPP specifies AoIP to use bandwidth-efficient AMR only;
however, in all known Osmocom world, our 2G BTS emit octet-aligned AMR
instead -- so in practice it makes most sense to run osmo-msc with
octet-align=1 as the default. osmo-msc shall at some point feature a
.cfg option to switch to octet-align=0 to conform with 3GPP, but until
then, osmo-msc now configures all AMR to be octet-align=1.
Depends: osmo-mgw If58590bda8627519ff07e0b6f43aa47a274f052b
Change-Id: Ief9225c9bcf7525a9a0a07c282ffb8cc0d092186
Looking at 3GPP TS 26.103 table 4.1, none of the AMR-Oxx variants are
supported by GERAN-GMSK nor UTRAN, so it makes no sense to include
these. There are no users of this.
Change-Id: I0cbc770fff55209676d9b6aae50011d6d5f897e2
(Same as osmo-bsc I47c9011b5e0e2886d221e34e6aa281d1dd0495c7)
*.vty tests are picked up by the Makefile.am by means of a wildcard --
they are run when they are there. So when you forget to add it to
EXTRA_DIST, it will be run in your local build tree, but it will be
silently omitted from a distribution tar, and nothing will complain
about it gone missing.
Instead, also use a *.vty wildcard in EXTRA_DIST. So any *.vty test
added to the git source will both be run *and* included in distribution
tars implicitly.
So far, test_neighbor_ident.vty was missing from the distribution.
Change-Id: Id28e020fc59b83d1b4cd0e5b72314a46bea62259
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
The comments indicating which two "members" are identical are
inaccurate. (One of them is a macro pointing at the other.)
Change-Id: Ifaa2f361db77cd0ed3ad39d6ca197195b9354ea1
Currently the CSD check is in the middle of figuring out the voice codec
for normal voice calls. Rather do the CSD check first, and then do voice
in one coherent section.
(prep for upcoming change in this code, to support AMR rate selection.)
Change-Id: Ibd21f0bb46c66a406904105564ce961a8760cbe7
Before the codec filter, it would have been the CN side codec, but now
it is only the codec that the RAN reports as assigned, fed into the
codecs filter.
(prep for upcoming change in this code, to support AMR rate selection.)
Change-Id: Ie7966099c5565013018734b0c2028484c24341a7
This also makes sure it doesn't compile against older libosmogsm gsup
versions which would break ABI.
Related: OS#6091
Depends: libosmocore.git Change-Id 70be3560659c58f24b8db529c4fc85da4bb0ec04
Change-Id: Ia002fd6e0334d56de34d352a0bf1a8604e2e9fd3
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
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
We do include this IE in result and error messages, but somehow
not in the request messages. For the sake of consistency, let's
ensure that the Source Name IE is present in all SMS related PDUs.
This additionally brings osmo-msc in sync with ttcn3-msc-test, which
was modified to expect the Source Name IE in all receive templates,
and makes the following testcases pass [again]:
* MSC_Tests.TC_gsup_mo_sms
* MSC_Tests.TC_gsup_mo_smma
* MSC_Tests.TC_gsup_mo_mt_sms_rp_mr
Change-Id: I65f5e3b7a0688e258979bb2679598659881a4321
Related: osmo-ttcn3-hacks.git Ic24d3082fe3dce08e43e8f3ecb6d6132503c55c6
Related: OS#6135
... so that it's clear which MNCC handler is used by looking
at the output of `show running-config`.
Change-Id: Id1fe7aecc1c8445db48ff5fddcf6df0f05ba5e2e
Prior to this change, if there was no explicit ipa-name configuration
in OsmoMSC, OsmoHLR would see the GSUP connection as MSC-00-00-00-00-00-00.
However, this default is constructed somewhere deep in IPA libraries
and is not visible to the GSUP client application, in this case OsmoMSC.
This situation creates a problem for SMS-over-GSUP routing: when we get
MT-forwardSM.req from an SMSC, we have to send a GSUP response, and this
response needs to get back to the MT-sending SMSC. Because OsmoHLR
applies only passive routing for these responses, we have to set
source_name when generating MT-forwardSM.res in OsmoMSC - but we cannot
do so if don't know our own IPA name.
Change the default OsmoMSC ipa-name from MSC-00-00-00-00-00-00 to
unnamed-MSC, mirroring OsmoHLR default of unnamed-HLR, and set it
at our application level rather than deep in the libraries.
Related: OS#6135
Change-Id: I7bacd001b81326c32bc262c7d0c0491ded822fa8
Previously added codecs tests uses non-default PT number sent by MT and
adopted by MO. Also test the other direction, i.e. a non-standard PT
from MO is adopted by MT.
Related: OS#6258
Change-Id: I8fbabe242982441d676d09f4d0ed7557c8349f2c
In msc_vlr_test_call.c, allow to tell MO non-default payload type
numbers in the SDP, to verify that it adopts the other call leg's PT
numbers.
Actually apply the non-default payload type number (AMR=96 instead of
the default of 112 from codec_mapping.c) and see the effects in
msc_vlr_test_call.err.
The diff shows that, as intended, the change in payload type number
should result in modifying the MGW endpoint to change the earlier '112'
to the modified '96' used in this test.
Related: OS#6258
Change-Id: I25df2ed7ad792fbe66dfd0fbf08182c9cf6cfc5b
In msc_vlr_test_call.c, allow to tell MO non-default payload type
numbers in the SDP, to verify that it adopts the other call leg's PT
numbers.
This test differs only slightly from the first codecs test, so in this
patch add the test as a 1:1 copy of the first test. The next patch [2/2]
will then show only the difference the new test makes.
Related: OS#6258
Change-Id: I618e3cf1b412985589a0c63bd76b7a60202f17b9
In patch I8760feaa8598047369ef8c3ab2673013bac8ac8a, osmo-msc learns to
handle codec mismatches reported by MT. For simplicity, that patch cuts
short the msc_vlr codecs tests by validating only the first codec.
Now test the full list of codecs properly.
This also introduces testing the re-assignment that MO does to match
MT's codec limitations, and removes the "EXPECTED FAILURE" markers.
Related: OS#6258
Change-Id: Ib933554f826c1b4347dfa3f6c4f6fe086be8b133
It was true once, but not since "do CN CRCX first"
Ie433db1ba0c46d4b97538a969233c155cefac21c
Related: OS#6258
Change-Id: I94e430e4e5b5bf18dbb155258d82f599ada453e6
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
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
We're checking the result of trans_alloc() 6 out of 7 times, so check
it in gsm_silent_call_start() too, for the sake of consistency.
Change-Id: Ie989cd8146d66d9531cf3f3d84f46a2c6fcc2e5c
Fixes: CID#322140
"[ESTABLISHED] transition to state ESTABLISHED not permitted"
i.e. don't complain when we already are in the established state.
Change-Id: I9b1fd63ed1ee7ed2877a4b2059386354598f4ea4
The cfg bits are for AMR-HR, not GSM-HR. The function
gsm0808_enc_speech_codec_list2() will return -EINVAL when it encounters
GSM-HR with non-zero cfg bits.
It appears this mapping was never used before, and my testing of call
re-assignment to match MT's codecs (it allows more than just the
assigned codec, because it can re-assign) has uncovered this bug
via MSC_Tests.TC_ho_inter_msc_out. I don't fully understand all the
details why we didn't see this before; anyway, the fix is obvious.
Change-Id: I19cff847a0f618ad000d12c1df54c55ef2f79699
The SGs interface is currently only casually mentioned in the chapter
running, even though the SGs interface is a prominent and often
requested feature. Let's give the SGs interface its own section so that
users can find the info about it quicker.
Related: OS#6008
Change-Id: Ic7c17511ee19cb7f6d5069b27beb661ecb4b0be8
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
Only the originator may terminate the VGCS/VBS call. This will not
happen in real life, because the UI of the MS should not allow
termination of a recevied VGCS call.
Change-Id: Ibe289920fa3ea50dd3e7d5c1371456dca9b72604
Related: OS#4854
Certain calls (seen on very old Nokias) won't have the rate adaptation flag
set on "analog" CSD calls. The field for the intermediate rate (after RA) is
still filled correctly.
Workaround this by setting the RA to V.110 whenever the RA is unset but an
intermediate rate is specified.
Change-Id: I5b3e5649fe071636f1becddfbfee06f9175a5f17
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
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
If the GSUP request message to which we are replying is an MT SMS
delivery from an SMSC relayed via OsmoHLR, we must set destination_name
in our reply - otherwise our reply won't make it back to the SMSC.
Related: OS#6135
Change-Id: I892fe87a733a78ed9d5761a8ce238caa135dea1e
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
This is a fixup for the patch
'3G: decapsulate IuUP to AMR at the MGW; allow 3G<-AMR->2G'
I386a6a426c318040b019ab5541689c67e94672a1
After above patch, osmo-msc intelligently decides which codecs to run on
which legs of the RTP streams. In the meantime, it seems the necessary
matching changes to call_leg_local_bridge() had been lost somehow.
Testing 3G to 3G voice now, I noticed that call_leg_local_bridge()
overwrites the intelligent choices made earlier.
The history of an MGW endpoint that should convert from IUFP to plain
AMR, extracted from a pcap, looks like this:
<- CRCX None None
-> CRCX-OK audio 4050 RTP/AVP 112 None
<- MDCX audio 4056 RTP/AVP 112 AMR
-> MDCX-OK audio 4050 RTP/AVP 112 AMR
<- MDCX audio 4056 RTP/AVP 96 VND.3GPP.IUFP
-> MDCX-OK audio 4050 RTP/AVP 96 VND.3GPP.IUFP
So after call_leg_local_bridge(), there is an extra MDCX + MDCX-OK that
switches the codec from 112 AMR back to 96 IUFP.
That is because call_leg_local_bridge() copies the *RAN* side's codec to
both CN sides, which used to be ok when RAN and CN codecs were always
identical.
Instead, adjust only the CN sides of the MGW endpoints, and adjust them
so that both CN sides are identical. osmo-mgw should then be able to
trivially translate the codecs appropriately.
Change-Id: I130bcd77ec57e332370c487a11b0b973b6e1089d
Neels Hofmeyr
Pau Espin
Vadim Yanitskiy
Andreas Eversberg
Philipp Maier
Manawyrm
Oliver Smith