Enable fmtp matching in sdp_audio_codecs_by_descr(). This has profound
effects on the codec_filter and other loops that collect codec matches.
By properly matching the meaning of fmtp, msc_vlr_test_call.c now works
as expected: all AMR variants are picked up properly. Adjust
expectations.
Change-Id: I63dc27e76243074ae0f019d219cae4412d322a84
So far the API is pivoting on the payload type numbers as the "primary
key" for codec lists. However, working with variants of codecs, the
payload type numbers are just incidental, and the API isn't helpful.
- change the behavior to regard the subtype name and fmtp as the
defining identity of a codec: use sdp_audio_codec_cmp() to match.
- add argument 'once': flag to make sure a given subtype+fmtp exists
only once, regardless of payload type nr.
- add argument 'pick_unused_pt_nr': flag to make sure a new entry
doesn't duplicate payload type numbers.
This is preparation to properly match AMR variants, in order to fix the
expected error currently visible in msc_vlr_test_call.c:875.
Change-Id: I87db779dbab39dfdef2724488ccdb6959e6731ed
In msc_a_up_call_assignment_complete(), iterate all the new AMR modes
and rates recently added to codec_mapping, and enlist all possible
matches.
Change-Id: I6163a51765efff998e05c9ee4e82a0a3759f2043
Use the full struct gsm0808_speech_codec instead of just the enum
gsm0808_speech_codec_type: allow testing specific AMR mode-sets on A/Iu
signalling.
Prepare for matching AMR fmtp parameters between call legs.
Change-Id: I387490525007bdf2e95306fd9b4e5e3b7e487d0f
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
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
The comments indicating which two "members" are identical are
inaccurate. (One of them is a macro pointing at the other.)
Change-Id: Ifaa2f361db77cd0ed3ad39d6ca197195b9354ea1
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
In msc_vlr_test_call, we fail to send the right MNCC struct for
MNCC_RTP_CREATE. We should pass a struct gsm_mncc_rtp. Fix that.
Change-Id: Ia0b3253f85c716e45f925da3f58f025af1f15ec9
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
Upcoming patch 'do CN CRCX first' changes the ordering of MGCP. To
properly show the change in behavior in the msc_vlr_test_call, first
clarify which side is expected to do MGCP when.
Related: SYS#5066
Change-Id: I972e7426006e5b62f81ccfe4fa224ee9eed7a7ac
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, 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
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
RANAP Security Command can include an encryption IE. If it includes
it the RNC can still ignore it (e.g. unsupported encryption) and
return the Security Command Complete with an choosen encryption IE:
"no encryption".
Validate the encryption element and ensure the encryption is included in
the encryption mask.
Closes: OS#4144
Change-Id: Icfc135c8b8ae862defe7114db492af600c26407f
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
In order for osmo-hlr to be able to 100% guarantee distinct INDs for CS and PS,
set CN-Domain = CS in all SendAuthInfo Requests.
In Milenage auth, it is highly desirable that osmo-hlr guarantees use of
distinct INDs for CS and PS domains. If an MSC and SGSN attached at the same
time use the same IND bucket to generate Milenage SQN, that collision would
rapidly waste SQNs and load osmo-hlr with requesting new auth tuples on each
CS/PS Complete-Layer3.
So far, osmo-msc did not indicate the CN domain in the GSUP SendAuthInfo
Request, which was neither required nor evaluated. The CN-Domain is only sent
for the UpdateLocation Request that usually follows later.
Related: OS#4318
Change-Id: I22f44068268e62801cadbf6542efaf153423cd65
Substantial parts of the CC / MNCC call establishment were so far completely
missing from the msc_vlr_test_call.c tests. With my new insights on CC and MNCC
procedures, complete the tests.
Root reason: since I am going to re-order the sequence of events to enable
codec negotiation via SDP in MNCC, I want to have comprehensive tests of the CC
procedures to see the effect as diffs in the test output.
Change-Id: Ie995e264eb1e3dd9558a1753ff6f9b55c1d084e1
Recently, the ability to run UTRAN without encryption was added, but the config
for it was tied to the A5 GERAN encryption configuration. This affected
osmo-msc's default behavior of Iu, breaking osmo-msc ttcn3 Iu tests: the ttcn3
test suite sets A5 to 0 (no encryption) but still expects Iu to enable air
encryption. Fix this "regression".
Add a separate vty config option for UEA encryption, even if it does not
provide full granularity to select individual UEA algorithms yet.
As a result, Iu default behavior remains to enable encryption regardless of the
A5 config. UTRAN encryption can be disabled by the new cfg option
"encryption uea 0" alone.
Even though the new vty command already allows passing various combinations of
the UEA algorithm numbers, only '0' and '1 2' are accepted as valid
combinations, to reflect current osmo-msc capabilities.
Revert most changes to the msc_vlr test suite in commit "do not force
encryption on UTRAN" (I04ecd7a3b1cc603b2e3feb630e8c7c93fc36ccd7): use new
net->iu_encryption instead of net->a5_encryption_mask.
Adjust/add to test_nodes.vty transcript tests.
Related: OS#4144
Change-Id: Ie138f2fcb105533f7bc06a6d2e6deccf6faccc5b
Remove the conditions that always enable encryption on UTRAN.
We so far lack an explicit configuration for UTRAN encryption, and this patch
does not add any either. Instead, whether UTRAN encryption is enabled is simply
triggered on whether GERAN has A5 encryption enabled (A5/n with n > 0). Though
GERAN and UTRAN encryption are not technically related at all, this makes UTRAN
behave like GERAN for now, until we implement a proper separate configuration
for UTRAN encryption.
Adjust the msc_vlr_test_* configuration by setting the net->a5_encryption_mask
such that the expected output remains unchanged. A subsequent patch
(I54227f1f08c38c0bf69b9c48924669c4829b04b9) will add more tests, particularly
cases of UTRAN without encryption.
Adjust manual and vty doc.
Related: OS#2783
Change-Id: I04ecd7a3b1cc603b2e3feb630e8c7c93fc36ccd7
GSM 04.08 10.5.4.11
The Release indication needs to have the Coding Standard set.
For phones that would display a message on screen, such as
"Number not in use", if the coding standard is not defined,
the display may show "Error in Connection"
Change-Id: Ib28b62a41d433e231cff5910d19455296b284df6
We also need stubs for the upcoming db_sms tests.
Due to a known bug of automake [1], we cannot use 'subdir-objects',
so as a side effect this change introduces some autoreconf warnings.
[1] https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=752993
Change-Id: I8846c940f2695fd33e1007fecac83e73f508bb34
While developing the inter-MSC handover refactoring, I was annoyed by the fact
that mncc_tx_to_cc() receives an MNCC message struct containing a msg_type, as
well as a separate msg_type argument, which may deviate from each other. So, as
a first step I wanted to make sure that all callers send identical values for
both by inserting an OSMO_ASSERT(msg_type == msg->msg_type). Later I was going
to remove the separate msg_type argument.
I then forgot to
- carry on to remove the argument and
- to actually test with internal MNCC (it so happens that all of our ttcn3
tests also use external MNCC).
As a result, the "large refactoring" patch for inter-MSC Handover breaks
internal MNCC operation.
Fix that: remove the separate msg_type argument and make sure that all callers
of mncc_tx_to_cc() indeed pass the desired msg_type in msg->msg_type, and hence
also remove the odd duality of arguments.
Various functions in mncc_builtin.c also exhibit this separate msg_type
argument, which are all unused and make absolutely no sense. Remove those as
well.
Related: OS#3989
Change-Id: I966ce764796982709ea3312e76988a95257acb8d
3GPP TS 49.008 '4.3 Roles of MSC-A, MSC-I and MSC-T' defines distinct roles:
- MSC-A is responsible for managing subscribers,
- MSC-I is the gateway to the RAN.
- MSC-T is a second transitory gateway to another RAN during Handover.
After inter-MSC Handover, the MSC-I is handled by a remote MSC instance, while
the original MSC-A retains the responsibility of subscriber management.
MSC-T exists in this patch but is not yet used, since Handover is only prepared
for, not yet implemented.
Facilitate Inter-MSC and inter-BSC Handover by the same internal split of MSC
roles.
Compared to inter-MSC Handover, mere inter-BSC has the obvious simplifications:
- all of MSC-A, MSC-I and MSC-T roles will be served by the same osmo-msc
instance,
- messages between MSC-A and MSC-{I,T} don't need to be routed via E-interface
(GSUP),
- no call routing between MSC-A and -I via MNCC necessary.
This is the largest code bomb I have submitted, ever. Out of principle, I
apologize to everyone trying to read this as a whole. Unfortunately, I see no
sense in trying to split this patch into smaller bits. It would be a huge
amount of work to introduce these changes in separate chunks, especially if
each should in turn be useful and pass all test suites. So, unfortunately, we
are stuck with this code bomb.
The following are some details and rationale for this rather huge refactoring:
* separate MSC subscriber management from ran_conn
struct ran_conn is reduced from the pivotal subscriber management entity it has
been so far to a mere storage for an SCCP connection ID and an MSC subscriber
reference.
The new pivotal subscriber management entity is struct msc_a -- struct msub
lists the msc_a, msc_i, msc_t roles, the vast majority of code paths however
use msc_a, since MSC-A is where all the interesting stuff happens.
Before handover, msc_i is an FSM implementation that encodes to the local
ran_conn. After inter-MSC Handover, msc_i is a compatible but different FSM
implementation that instead forwards via/from GSUP. Same goes for the msc_a
struct: if osmo-msc is the MSC-I "RAN proxy" for a remote MSC-A role, the
msc_a->fi is an FSM implementation that merely forwards via/from GSUP.
* New SCCP implementation for RAN access
To be able to forward BSSAP and RANAP messages via the GSUP interface, the
individual message layers need to be cleanly separated. The IuCS implementation
used until now (iu_client from libosmo-ranap) did not provide this level of
separation, and needed a complete rewrite. It was trivial to implement this in
such a way that both BSSAP and RANAP can be handled by the same SCCP code,
hence the new SCCP-RAN layer also replaces BSSAP handling.
sccp_ran.h: struct sccp_ran_inst provides an abstract handler for incoming RAN
connections. A set of callback functions provides implementation specific
details.
* RAN Abstraction (BSSAP vs. RANAP)
The common SCCP implementation did set the theme for the remaining refactoring:
make all other MSC code paths entirely RAN-implementation-agnostic.
ran_infra.c provides data structures that list RAN implementation specifics,
from logging to RAN de-/encoding to SCCP callbacks and timers. A ran_infra
pointer hence allows complete abstraction of RAN implementations:
- managing connected RAN peers (BSC, RNC) in ran_peer.c,
- classifying and de-/encoding RAN PDUs,
- recording connected LACs and cell IDs and sending out Paging requests to
matching RAN peers.
* RAN RESET now also for RANAP
ran_peer.c absorbs the reset_fsm from a_reset.c; in consequence, RANAP also
supports proper RESET semantics now. Hence osmo-hnbgw now also needs to provide
proper RESET handling, which it so far duly ignores. (TODO)
* RAN de-/encoding abstraction
The RAN abstraction mentioned above serves not only to separate RANAP and BSSAP
implementations transparently, but also to be able to optionally handle RAN on
distinct levels. Before Handover, all RAN messages are handled by the MSC-A
role. However, after an inter-MSC Handover, a standalone MSC-I will need to
decode RAN PDUs, at least in order to manage Assignment of RTP streams between
BSS/RNC and MNCC call forwarding.
ran_msg.h provides a common API with abstraction for:
- receiving events from RAN, i.e. passing RAN decode from the BSC/RNC and
MS/UE: struct ran_dec_msg represents RAN messages decoded from either BSSMAP
or RANAP;
- sending RAN events: ran_enc_msg is the counterpart to compose RAN messages
that should be encoded to either BSSMAP or RANAP and passed down to the
BSC/RNC and MS/UE.
The RAN-specific implementations are completely contained by ran_msg_a.c and
ran_msg_iu.c.
In particular, Assignment and Ciphering have so far been distinct code paths
for BSSAP and RANAP, with switch(via_ran){...} statements all over the place.
Using RAN_DEC_* and RAN_ENC_* abstractions, these are now completely unified.
Note that SGs does not qualify for RAN abstraction: the SGs interface always
remains with the MSC-A role, and SGs messages follow quite distinct semantics
from the fairly similar GERAN and UTRAN.
* MGW and RTP stream management
So far, managing MGW endpoints via MGCP was tightly glued in-between
GSM-04.08-CC on the one and MNCC on the other side. Prepare for switching RTP
streams between different RAN peers by moving to object-oriented
implementations: implement struct call_leg and struct rtp_stream with distinct
FSMs each. For MGW communication, use the osmo_mgcpc_ep API that has originated
from osmo-bsc and recently moved to libosmo-mgcp-client for this purpose.
Instead of implementing a sequence of events with code duplication for the RAN
and CN sides, the idea is to manage each RTP stream separately by firing and
receiving events as soon as codecs and RTP ports are negotiated, and letting
the individual FSMs take care of the MGW management "asynchronously". The
caller provides event IDs and an FSM instance that should be notified of RTP
stream setup progress. Hence it becomes possible to reconnect RTP streams from
one GSM-04.08-CC to another (inter-BSC Handover) or between CC and MNCC RTP
peers (inter-MSC Handover) without duplicating the MGCP code for each
transition.
The number of FSM implementations used for MGCP handling may seem a bit of an
overkill. But in fact, the number of perspectives on RTP forwarding are far
from trivial:
- an MGW endpoint is an entity with N connections, and MGCP "sessions" for
configuring them by talking to the MGW;
- an RTP stream is a remote peer connected to one of the endpoint's
connections, which is asynchronously notified of codec and RTP port choices;
- a call leg is the higher level view on either an MT or MO side of a voice
call, a combination of two RTP streams to forward between two remote peers.
BSC MGW PBX
CI CI
[MGW-endpoint]
[--rtp_stream--] [--rtp_stream--]
[----------------call_leg----------------]
* Use counts
Introduce using the new osmo_use_count API added to libosmocore for this
purpose. Each use token has a distinct name in the logging, which can be a
globally constant name or ad-hoc, like the local __func__ string constant. Use
in the new struct msc_a, as well as change vlr_subscr to the new osmo_use_count
API.
* FSM Timeouts
Introduce using the new osmo_tdef API, which provides a common VTY
implementation for all timer numbers, and FSM state transitions with the
correct timeout. Originated in osmo-bsc, recently moved to libosmocore.
Depends: Ife31e6798b4e728a23913179e346552a7dd338c0 (libosmocore)
Ib9af67b100c4583342a2103669732dab2e577b04 (libosmocore)
Id617265337f09dfb6ddfe111ef5e578cd3dc9f63 (libosmocore)
Ie9e2add7bbfae651c04e230d62e37cebeb91b0f5 (libosmo-sccp)
I26be5c4b06a680f25f19797407ab56a5a4880ddc (osmo-mgw)
Ida0e59f9a1f2dd18efea0a51680a67b69f141efa (osmo-mgw)
I9a3effd38e72841529df6c135c077116981dea36 (osmo-mgw)
Change-Id: I27e4988e0371808b512c757d2b52ada1615067bd
Add an SGs interface (3GPP TS 29.118) to osmo-msc in order to support
SMS tunneling and Circuit Switched Fallback (CSFB)
Change-Id: I73359925fc1ca72b33a1466e6ac41307f2f0b11d
Related: OS#3615
Replace locally defined enum ran_type with libosmocore's new enum
osmo_rat_type, and value_string ran_type_names with osmo_rat_type_names.
The string representations change, which has cosmetic effects on the test suite
expectations.
Depends: I659687aef7a4d67ca372a39fef31dee07aed7631 (libosmocore)
Change-Id: I2c78c265dc99df581e1b00e563d6912c7ffdb36b
Always use LAC which is part of Cell Global ID otherwise we might end up
in a situation where separately stored LAC differs.
Both are described in 3GPP TS 23.008 $2.4 as temporary subscriber data
to be stored in VLR. Both are defined in 3GPP TS 23.003. The LAC is part
of LAI which is part of CGI so there should be no case when those values
differ for a given subscriber.
Change-Id: I993ebc3e14f25e83124b6d3f8461a4b18f971f8e
Avoid leaking details on accessing data structure for LAC value into
test output: that's irrelevant clutter which forces unnecessary test
output modifications.
Change-Id: I4a1d7884cf47ad513d7d6fb27c5c6f1b829dff2e
Following previous rename of gsm_subscriber_connection:
Some functions and #defines are still called like "msc_conn" or just "msc_",
while they are clearly about a RAN conn.
To avoid confusion with the future separate concepts of MSC roles and a RAN
connection, rename all those to match the common "ran_conn" prefix.
Change-Id: Ia17a0a35f11911e00e19cafb5d7828d729a69640
Neels Hofmeyr
Alexander Couzens
Vadim Yanitskiy
Keith Whyte
Harald Welte