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
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
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
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
Also regard an RTP port as invalid if the IP address is 0.0.0.0.
Achieve this by using osmo_sockaddr_str_is_nonzero() instead of
osmo_sockaddr_str_is_set().
Depends: I73cbcab90cffcdc9a5f8d5281c57c1f87b2c3550 (libosmocore)
Change-Id: I53ddb19a70fda3deb906464e1b89c12d9b4c7cbd
libosmo-mgcp-client recently introduced osmo_mgcpc_ep_cancel_notify() to cancel
notification if a notify target FSM deallocates. Use it for sanity in
rtp_stream FSM cleanup, the notify target for endpoint FSMs.
Depends: I41687d7f3a808587ab7f7520f46dcc3c29cff92d (osmo-mgw)
I14f7a46031327fb2b2047b998eae6ad0bb7324ad (osmo-mgw)
Change-Id: I351bb8e8fbc46eb629bcd599f6453e2c84c15015
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
Neels Hofmeyr
Andreas Eversberg
Vadim Yanitskiy
Pau Espin