This fixes a bug in the following circumstances:
* BSIC is set to 0 in the config file
* No TSC is explicitly specified at the BST level in the config file
In this case, we ended up using BSIC=0 and TSC=7, as TSC=7 is our
default initialization value.
The TSC of the CCCH/BCCH must always be the BCC, which is the lower 3
bits of the BSIC. Having configuration options for both the BSIC _and_
the TSC at the BTS level therefore makes no sense, as it only adds ways
in which users can configure non-oprational configurations. So we
remove the bts->tsc member, and keep only the ts->tsc members that allow
us to configure a timeslot-specific TSC that's different from the BTS
TSC (= BCC).
Fit most of the code in 80 chars width. Some instances still leak past 80
characters because of long function names, inline comments or the like, "the
exception proves the rule."
Sponsored-by: On-Waves ehi
Allow logging the plane (Ctrl/User) and side (SGSN/GGSN) in functions that only
have a gtphub_bind* to work with, by adding a constant label to each bind.
Sponsored-by: On-Waves ehi
Initialize llist_heads to empty (2 were missing). Move those for struct gtphub
instances to gtphub_zero() (one moved, one added).
In from_[gs]gsns_read_cb(), use a return type that can actually reflect
negative return values.
resolved_addr.buf: no need to take the address of a byte array var
(cosmetic).
Pass the proper user data address to sgsn_ares_query(), not the address of
the pointer holding the user data address.
Initialize ggsn_lookup->expiry_entry (was missing). Publish the function for that
in gtphub.h so gtphub_ext.c can use it.
Sponsored-by: On-Waves ehi
Generalize to make the PDP ctx message definitions and "sending" of messages
from SGSN->gtphub->GGSN and back reusable in future tests.
Publish gsn_addr_from_sockaddr() in gtphub.h for use in gtphub_test.c.
Use an osmo_sockaddr for resolved_ggsn_addr, because one is needed for
comparison in probably every future test.
Add LVL2_ASSERT() to print assertion message and return instead of abort,
so that functions can be called from several tests without losing the
info of which test caused it from which line.
Use globals for struct gtphub and time_t now, to reduce nr of args that need to
be passed around when writing tests. Add a default test setup function.
Sponsored-by: On-Waves ehi
Up to now I used the Echo as a test for sequence nr mappings. But Echos
should be handled differently: they are scoped on the link and an Echo
response should be sent right back to the requester.
Sponsored-by: On-Waves ehi
For the resolving function, change the function signature to return a
gtphub_peer_port. In consequence, publish two functions concerned with
gtphub_peer_port instances for use in test and gtphub_ext.c.
Add GGSN resolution queue, callback and cache. Simple implementation: if an
SGSN asks for a GGSN, it will first get no answer, and I hope it will ask again
once the GGSN is in the cache.
Within gtphub_ext.c, have a dummy sgsn struct, as the sgsn_ares code currently
depends on it (half the functions pass an sgsn instance pointer around, but the
other half use the global one).
In the unit tests, wrap away the ares initialization so that they can work
without a DNS server around. The netcat test breaks because of this, will
remove it.
Using sgsn_ares, implement the gtphub_resolve_ggsn_addr() function, I hope:
untested.
Minor cosmetics just to see if you're paying attention... ;)
Sponsored-by: On-Waves ehi
First steps towards a new GTP hub. The aim is to mux GTP connections, so that
multiple SGSN <--> GGSN links can pass through a single point. Background:
allow having more than one SGSN, possibly in various remote locations.
The recent addition of OAP to GSUP is related to the same background idea.
(This is a collapsed patch of various changes that do not make sense to review
in chronological order anymore, since a lot of it has thorougly transmorphed
after it was first committed.)
Sponsored-by: On-Waves ehf
Extend the ul/dl counting to count the usual messages on the
Gb interface. Add counters for the attach, routing area update,
pdp context activation and deactivation procedures. Update the
test result with the new counters.
Trigger an OAP registration upon IPA connect. Feed incoming OAP messages to
oap_handle() and send replies returned by it.
Add oap_config to sgsn_config (todo: vty).
Sponsored-by: On-Waves ehf
[hfreyther: Fix coding style]
Add new kitchen sink openbsc/utils.h and libcommon/utils.c to make three so far
static functions public (so I can use them in the upcoming OAP code).
A place to put them could have been the gprs_utils.h, but all general functions
in there have a gprs_ prefix, and todo markings to move them away. All other
libcommon headers are too specific, so I opened up this kitchen sink header.
Replace the implementation of encode_big_endian() with a call to
osmo_store64be_ext(). See comments.
Apply the change in Makefiles and C files.
This change has some implications for the test case. It manipulated
bss_ptmsi_state and sgsn_tlli_state variables to make the output of
rand_r() and thus the TLLI/TMSI used predictable.
This possibility is gone when using RAND_bytes() so instead it is
overridden by a function that returns a deterministic sequence of values
(0x00dead00, 0x00dead01, ...). The test cases are adapted to expect
these values instead of the pseudo random values before.
The gbproxy_test stdout file changes as well, but only where the
TLLI/TMSI is displayed (in the hex dumps as well as the TLLI cache
entries). All other output is the same.
This API is a bit unfortunate as the caller will also
access the endpoint directly. E.g. like this:
output = bsc_mgcp_rewrite(...,
mgcp_net_src_addr(endp),
endp->net_end.local_port, -1,
In terms of "terminology" the "net" was meant to be bad
internet and the "bts" is the local and trusted network
segment. With this terminology the "bts" would be the
call-agent/MGW and "net" where the BSCs will send data
to but that is not the case and terminology actuallys
refers to:
* net: The addresses exposed to the entity that
made the MGCP call
* bts: The system where we get our data for the
local audio flow.
Fix the method but leave the API as it is. Use the net_end
in the net_src method and the bts_end in the bts_src method.
There appears to be a leak of CIDs:
<000b> mgcp_osmux.c:544 All Osmux circuits are in use!
There are paths that a CID had been requested and never released
of the NAT. Remember the allocated CID inside the endpoint so it
can always be released. It is using a new variable as the behavior
for the NAT and MGCP MGW is different.
The allocated_cid must be signed so that we can assign outside
of the 0-255 range of it.
Fixes: OW#1493
Some systems only want to use Osmux. In case only Osmux
should be used fail if it has not be offered/acked.
Client:
Verified On, Off and Only with X-Osmux: 3 and without this field.
<000b> mgcp_protocol.c:823 Osmux only and no osmux offered on 0x14
<000b> mgcp_protocol.c:884 Resource error on 0x14
NAT:
Not tested and implemented
Fixes: OW#1492
* Test that one can get an id
* That they are assigned predicatble right now
* That returning them will make the number of used ones go down
* That allocating more will fail
The signature of mr_config and the BSC implementation didn't
match and the compiler was warning about it:
osmo_bsc_api.c:530:2: warning: initialization from incompatible pointer type
.mr_config = bsc_mr_config,
^
osmo_bsc_api.c:530:2: warning: (near initialization for ‘bsc_handler.mr_config’)
Change the mr_config again and provide an implementation
that will set the ms and bts data structure. It would be
better to put the size outside of the IE but I am not going
to change it right now. It would also be nice to either move
the AMR setting into the "nitb" structure or have the msc
data be used _after_ the bts settings. This needs to be
cleaned up in the next step.
Manually verified by placing a MO call and checking that
both the channel mode modify and the mode modify request
contain the multi rate config with the rate mr config
(length two bytes, version 1, icmi==1, no start mode being
set).
This way a lot of if/else can just be killed by the caller deciding
which of the two instances to use.
I have copied both branches to new files, replace bts for ms in one
of them and ran diff on it. There is no difference.
In case of the RTP bridge mode we need to select the codec
ourselves. Rely on the same (incomplete) codec selection that
can be done using the mncc-int configuration node. This might
gain bearer capabilities support.
In case of a SDCCH a TCH/F will be attempted to be assigned.
This is an open issue for both modes and there should be a
preference for full or half-rate channels somewhere.
Implement sending MDCX on the newly allocated channel and send
the data to the same destination as the currently connected one.
This way the receiver can implement RTP RFC Appendix A.1 and
deal with the new source.
For the LCR rtp-bridge audio should directly flow to the
remote system. In contrast to the original patch audio
will now flow directly from the BTS to the remote system.
This assumes that BTS and the remote system are in the
same network segment and can directly communicate.
There are various limitations in the first iteration of
the implementation:
We could (and in the future) should delay the assignment
but currently we are forced to pick the channel and move
it to the audio state. In case we are located on a SDCCH
we always need to change but if we are on a TCH we could
send the ipa.CRCX and change the audio state a lot later.
The net effect is that the audio codec selection needs to
be done in the NITB code and not in the system connected
to it.
This only works with ip based systems. For E1 systems one
could still use the RTP socket or even try to move this
out of the process.
There is no code for handover handling and it relies on
the remote system dealing with the SSRC change of the
system.
This adds the protocol definition for the RTP bridge extension
of Andreas Eversberg and bumps the protocol version.
I added the missing mncc mappings from value to string.
[ 5cf8fb10ea3addcae74d37f4dbf1c1be664df53e protocol extension
5dac90de38990b188f499c602bf18a4f232070e8 payload extension]
When using multiple interfaces on a system one can now configure
which will be served for the BTS ports and which will be served
for the network. The direct usage of source_addr is now only to
initialize the MGCP receiving port itself.
Make it possible to bind the call-agent to a specific IP address
and the network and bts end to different ip addresses. Begin by
clarifying which source ip address we want to have.
Use the existing ulaw encode/decode to support PCMU as well.
The MERA VoIP switch has some severe issues with the GSM codec
and it appears easier to enable transcoding for it.
The mera switch doesn't appear to cope with codec change
between a SIP 180 trying and the 200 ok connection result.
Inserting the codec is touching too many places. Ideally we
should have the transcoding function as pointer in the struct
as well but the arguments differ.. so it is not a direct way
forward.
Iridium is a satellite network which operates a GPRS-like that allows you to
get speeds up to 128kbit/s. However, it takes from 5 to 6 secs to get the
bandwidth allocated, so the conversation is garbled during the time.
This patch uses the new dummy padding support in libosmo-netif that is
controlled through the osmux osmux_xfrm_input_open_circuit().
This includes a new VTY option for osmux.
We have a lot of legacy that I am afraid to break. We have
everything in place to make a good codec selection (e.g. if
we can avoid transcoding, pick the one with best quality or
the lowest speed). Right now I have a specific case where
from all options I want to pick GSM. Guard the codec compat
check behind the disallow transcoding option to make sure
to not break legacy application.
The SDP file handling will get more complicated in terms of
codec selection so let's remove it from the protocol handling
before we start blowing it up in size.
The parsing code assumed that there will be a single payload
type and this assumption is clearly wrong. Forward all of the
payload types. The code is still only extracting the first
type from the list. The variable name has been renamed to
reflect this.
In case foreign simcards are used we can not do authentication
and ciphering. In case a TMSI is re-used too early and we do
page using TMSI we can't know which of the two MS is responding
to us. We could change the "secure channel" routine to ask for
the IMSI and only then stop the paging.
As we don't have ciphering there is not much use in using the
TMSI. Add a mode "no assign-tmsi" that will not assign the TMSI
during LU. Now CM Service Request and Paging Response will
work using the IMSI. There can't be a clash with that.
[ciaby fixed the vty write to use the right name]
in_address is not 'accidently' included by FreeBSD when we include
the osmocom/core/select.h header file. We need to include a bit
more.
In file included from mgcp_protocol.c:38:
../../include/openbsc/mgcp_internal.h:134:21: error: field has incomplete type 'struct sockaddr_in'
struct sockaddr_in forward;
Struct osmo_msc_data contains int core_ncc, which is actually the
MNC part of the PLMN, not to be confused with the Network Colour
Code.
The following patch renames this field for clarity and consistency
with the standards.
We don't need to consume all the entropy of the kernel but can
use libcrypto (OpenSSL) to generate random data. It is not clear
if we need to call RAND_load_file but I think we can assume that
our Unices have a /dev/urandom.
This takes less CPU time, provides good enough entropy (in theory)
and leaves some in the kernel entropy pool.
We are using the token to find the right bsc_config and
then we can use the last_rand of the bsc_connection to
calculate the expected result and try to compare it with
a time constant(???) memcmp.
Check if the NAT has sent 16 bytes of RAND and if a key
has been configured in the system and then generate a
result using milenage. The milenage res will be sent and
noth the four byte GSM SRES derivation.
Generate 16 byte of random data to be used for A3A8 by
the BSC in the response. We can't know which BSC it is
at this point and I don't want to send another message
once the token has been received so always send the data
with an undefined code. The old BSCs don't parse the
message and will happily ignore the RAND.
/dev/urandom can give short reads on Linux so loop
around it until the bytes have been read from the kernel.
Instead of doing open/read/close all the time, open the
FD in the beginning and keep it open. To scare me even
more I have seen /dev/urandom actually providing a short
read and then blocking but it seems to be the best way
to get the random byes we need for authentication.
So one should/could run the cheap random generator on
the system (e.g. haveged) or deal with the NAT process
to block.
Unfortunately the basic structure of the response is broken.
There is a two byte length followed by data. The concept of
a 'tag' happens to be the first byte of the data.
This means we want to write strlen of the token, then we
want to write the NUL and then we need to account for the
tag in front.
Introduce a flag if the new or old format should be used.
This will allow to have new BSCs talk to old NATs without
an additional change. In the long run we can clean that up.
The code to do that doesn't belong to the control interface, so
abstract it out to a separate function gsm_bts_set_system_infos().
[hfreyther: Fix the coding style...]
For real networks we need to check if the requested APN string
is allowed and then resolve the GGSN address through DNS. There
are countries with two or three digit MNCs and one could either
try to keep a list of countries that have two/three digits or
just try both of them. I have opted for the later for the ease
of the implementation.
C-Ares doesn't allow to cancel a request so we will need to
have the MMCTX and the Lookup have different lifetimes. We simply
set ->mmctx to NULL in case the MMCTX dies more early.
The selected and verified apn_str will be copied into the out
parameter. In case no static APN/GGSN config is present and the
dynamic mode is enabled a request will be made.
c-ares is an asynchronous DNS resolver and we need it to
resolve the GGSN address. This is integrating the library
into our infrastructure. We will create and maintain a list
of registered FDs (c-ares is currently only using one of
them) and (re-)schedule the timer after events occurred.
Harald Welte
Neels Hofmeyr
Holger Hans Peter Freyther
Daniel Willmann