The log message does not help and says where the data is
being sent to. This is because we have both a RTP and RTCP
port. Remember if we failed with RTCP or RTP and improve
the log message.
I was searching a case where the port was bound to a local
address (e.g. 127.0.0.1) and tried to send the data to a
public one (e.g. 8.8.8.8).
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.
Merge two copies into a local static helper function. The format
of the message will change and then it is easier to modify it in
one place than in two.
Sadly the original patch was merged before this clean-up so do
the clean-up as second step.
Conflicts:
openbsc/src/libbsc/abis_rsl.c
openbsc/src/libbsc/gsm_04_08_utils.c
The pre-release didn't add a newline after the apn and the patching
pattern command. Create a quirk command that combines both. The
pre-release didn't include a differentation between routing and
patching.
The TLLI handling has a different and more generic name now. Make
it handle the old one that is actively used.
Add a file with the broken format and the standard config file
test should pick it up.
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]
Mike's patch included clean-ups I want to apply separately and
change them a bit. If we return from an else we don't need to
put the else.
* Try the E1 trunk first
* Then try a local virtual trunk
* Fail if none of the above returned
Remove the host portion of the endpoint Id. This requires less
configuration and we are probably fine to trust that MGCP only
received messages designated for it.
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.
I guess none of our users knows what a mi_type=0x02 is, but most would
know what an IMSI or a TMSI is. So let's use the newly introduced
gsm48_mi_type_name() function to fix this.
For some odd reasons the XID is not a separate SAPI but has been kludged into
the GMM SAPI. This means we ahve to be careful not to dispatch XID frames into
GMM. We do this by introducing an explicit check for UI frames before the
dispatch to GMM.
The previous code already was doing "the right thing" but printed occasional
messages like "gprs_gmm.c:2082 Unknown GSM 04.08 discriminator 0x01: 01 00 0e
00 32 11 03 16 01 90 63 28 0b". Those should be gone after this patch.
Traffic cannot sent to BTS, if there is (currently) no logical channel
associated with the transaction.
This happens, if TCH traffic is received from upper layer, but there is
no lchan available before completing immediate assignment, handover or
assignment process.
[hfreyther: The code has not been moved to tch_frame_down
but the issue looks similiar]
The NAT sends an incomplete SDP file for the purpose of informing
the BSC about the remote IP/PORT early. The case of an incomplete
SDP file was not considered. Check if there is a codec and if not
skip it.
TODO: We need to have a better end-point life cycle test.
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.
First collect everything we know and the mapping. E.g. a genuis
could remap "3" to "AMR" so we only know the codecs once we are
at the end of the SDP file. Once we have collected everything we
can select the audio codecs. The current code is compatible in
that two codecs will be selected regardless of if they make any
sense or not.
mgcp_set_audio_info could re-use some of our codec information
but then the caller in the MGCP protocol needs to be updated as
well as we use the "I: GSM" information to derive the codec from
there.
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.
Using the talloc leak report we see that there are some msgb's
that are allocated for SMS but we don't have transactions or
SMS around. We need to improve the name of the messages to
uniquely dscribe where they are from but the obvious leak does
occur in this routine.
The no available transaction id is most likely the case where
we leak memory. This should not occur and shows another issue
with the smsqueue/smpp handling. It doesn't explain the subscr
reference count issue either.
Extract of the leak report:
GSM 04.11 contains 1160 bytes in 1 blocks (ref 0) 0x2517dc0
GSM 04.11 contains 1160 bytes in 1 blocks (ref 0) 0x24b56e0
GSM 04.11 contains 1160 bytes in 1 blocks (ref 0) 0x23e7930
In case the subscriber is currently busy we would omit the
subscr_put. This seems to be very hard to hit as the subscr
need to be active and at the same time be selected for the
purge operation.
As the comment says we should not rely that the paging
occurs on the current LAC. We might page at more BTS.
Walk all the BTS to stop paging. No callbacks will be
issued by this stop operation.
In case we can't page on a BTS then stop it everywhere. The
callers of paging_request assume that this is kind of an
atomic operation and we should help with that.
Coordinate with the normal subscriber channel requests instead
of going to page ourselves. This might lead to getting a channel
that is of a different type though.
vty_interface_layer3.c:584:4: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long unsigned int' [-Wformat=]
sizeof(subscr->extension)-1, VTY_NEWLINE);
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]
When we can't find the TMSI then the subscriber is not in our
VLR. We have not consulted with the HLR and it is better to not
use such a severe error code.
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.
If we have tried SMPP first and it was not routable, and then
tried the local delivery there is no point in trying SMPP with
the same parameters again. Leave early and return unknown sub
to the caller.
default-route would only be looked at after there has been
no subscriber in the local database. Depending on the setup
this is not what one wants. This has been discussed at the
OsmoDevCon and there have been hacks in some branches. Let's
introduce a VTY command to select if SMPP should be consulted
first and then fallback to the current behavior.
Even if it is using BSC/NITB types let's put it in the header
file than just declaring it at a place that could bitrot in a
way that doesn't lead a warning.
The "default-route" for SMPP will be used after a local
subscriber look-up. Sometimes we want to route everything
to SMPP. Make this possible by changing this routine.
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.
In case the token was not correct, just close the connection.
It is not clear that forcing a new TCP connection is going to
give us any extra security here. But with the upcoming auth
handling it does make sense to have both case look similar.
In the libfilter source code, which is built regardless of --enable-nat,
headers from libosmo-sccp were used, thus causing a build failure (see
below) when building without --enable-nat, and libosmo-sccp not being
installed (or being installed in a prefix not otherwise included in the
build).
The build fails like this:
In file included from ../../../src/libfilter/bsc_msg_filter.c:27:0:
../../../include/openbsc/bsc_nat_sccp.h:27:37: fatal error: osmocom/sccp/sccp_types.h: No such file or directory
As the includes seem not to be actually needed, this change fixes the
issue by just omitting them.
There was no context for the SCCP CREF message and this means
that the msc_con was a plain NULL pointer that was dereferenced
and the application would crash.
Use the new API to pass the incoming MSC Connection which sould
be used for the SCCP CREF message as context. The code has not
been fed with an actual SCCP CR message.
The loop was used to print all returned addresses but we can
simply pick the first one. This is fixing a coverity issue that
the loop will be executed eaxactly once (and that was on
purpose).
Simplify the code and just take the first element (which might
be NULL).
Fixes: Coverity CID#1302852
We can't do much in case the fd is failing to be registered.
There should be a timeout that is catching this and it might
be able to repair it self.
Fixes: Coverity CID#1302854
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...]
In case the query for "hostname" will fail c-ares will append the
domain name of /etc/resolv.conf and query again. We don't want that
so claim we provide a list of domain names and then don't provide
any.
I didn't intend to have pushed the c-ares code to master yet.
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.
When needing to do an asynchronous DNS query we need
to keep the TLV data around. So create a wrapper that
takes a copy of it and frees it after the call. I can
change the code to add an out parameter to decide if
the msgb should be freed or not.
Pick network failure in case the msgb could not be
cloned in the hope the MS will retry then.
A real SGSN will dynamically resolve the APN name into the
GGSN IP Address. This means that after we have collected all
information we need to start to resolve the GGSN and then
can continue.
This is a left-over from the initial system where no PDP
was provided by the system. For now if there is a subscr
attached and no PDP context provisioned. He is not allowed
to have a data connection.
Update the testcase to create the pdp list entry more
early with a wildcard and then change it to a specific
match.
Include the hlr-Number of the subscriber in the CDR. This is useful
for debugging and understanding which equipment was used during the
test. In contrast to the MSISDN the '+' is emitted as the number
must be in international format already.
Copy the hlr-Number into the sgsn_data and use it during
the purgeMS. There is no unit test that looks at the data
we send so I manually verified this by looking at the output.
Below is the output of the test that purges the subscriber.
<000f> gprs_subscriber.c:170 SUBSCR(123456789012345) Sending GSUP, will send: 0c 01 08 21 43 65 87 09 21 43 f5 09 07 91 83 61 26 31 23 f3
We have verified/selected the APN. Either based on the subscriber
data, a global APN match. But at least this SGSN has looked at
what the MS has asked for and then selected a matching GGSN.
Daniel Willmann
Holger Hans Peter Freyther
Neels Hofmeyr
Harald Welte