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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
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Extend the osmux only setting from the MGCP MGW to the NAT. This
is applied when an endpoint is allocated and/or when the allocation
is confirmed by the remote system.
Not tested. The impact should only be when the new option is
being used.
Fixes: OW#1492
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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
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* Print number of used CIDs for the system
* Hopefully this is just the beginning
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sizeof(uint8_t) == 1 and there is no need to create an array
with 16 bytes and then only use the first two of them. This
means the CID range is from 0 to 127 and we should be able
to extend this to 256 by changing the array size to 32. Update
the testcase now that we can have more than 16 calls with Osmux.
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* 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
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[hfreyther: Add cast to uint8_t to fix compiler warning]
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[hfreyther: Fix compiler warning about unused variable]
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[hfreyther: Link to libcrypto, include header, add uint8_t* cast]
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Require openssl version to be >= 0.9.5 because we rely on the RAND_bytes
return value.
[hfreyther: Add cast to uint8_t*]
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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).
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Sponsored-by: On-Waves ehf
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Move IPA header composition and sending to new static gsup_client_send() (so I
can use it in the upcoming OAP code).
Sponsored-by: On-Waves ehf
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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).
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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.
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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
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clang complained that different enums are mixed with the
return type and we actually want this to be an int now.
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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.
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Make every line use same punctuation style ('.' at the end and start in caps).
Insert spaces to have all help texts in a column.
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The build process requires pcap/pcap.h in openbsc/src/utils/meas_pcap2db.c, but
the configure script did not check for that file.
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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.
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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.
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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.
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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]
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Before:
<command id='osmux dummy (on|off)'>
<params>
<param name='osmux' doc='RTP multiplexing' />
<param name='dummy' doc='Enable dummy padding' />
<param name='on' doc='Disable dummy padding' />
<param name='off' doc='(null)' />
</params>
After:
<command id='osmux dummy (on|off)'>
<params>
<param name='osmux' doc='RTP multiplexing' />
<param name='dummy' doc='Dummy padding' />
<param name='on' doc='Enable dummy padding' />
<param name='off' doc='Disable dummy padding' />
</params>
</command>
Note the 'null' string in 'off'. Reported by Holger.
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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
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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.
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In the 1900 MHz PCS band, we always generated neighbor cell lists
consisting of only a single neighbor cell, rather than al the
configured BTSs.
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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.
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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.
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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.
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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.
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The spec unfortuantely uses both terms and has no consistent spelling,
but in our logging output we can at least try to be consistent.
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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.
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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.
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The same radio link timeout value is used for BTS and MS side.
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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]
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