Subject: openbsc.git branch fairwaves/testing created. 0.13.0-394-g7b37e1f
From: [email-masked]
To: [email-masked]
Date: Tue, 15 Apr 2014 13:26:30 GMT

This is an automated email from the git hooks/post-receive script. It was
generated because a ref change was pushed to the repository containing
the project "The OpenBSC GSM Base Station Controller (+MSC/HLR/SGSN)".

The branch, fairwaves/testing has been created
        at  7b37e1f8c63fa2315557d47aecfdb2e3cfc10b18 (commit)

- Log -----------------------------------------------------------------
http://cgit.osmocom.org/openbsc/commit/?id=7b37e1f8c63fa2315557d47aecfdb2e3cfc10b18

commit 7b37e1f8c63fa2315557d47aecfdb2e3cfc10b18
Author: Andreas Eversberg <[email-masked]>
Date:   Thu Mar 7 15:25:48 2013 +0100

    HACK: Fix of several compiler warnings

http://cgit.osmocom.org/openbsc/commit/?id=006b7fa9107d70e9686da2baef0e0166ae9150e4

commit 006b7fa9107d70e9686da2baef0e0166ae9150e4
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Mar 31 12:01:25 2013 +0200

    osmo-nitb support for codec negotiation
    
    The caller's most preferred codec is selected out of the union of codecs,
    which both parties support.
    
    Since codec negotiation is done automatically, there is no need to define
    codec for TCH/F and TCH/H via VTY anymore.

http://cgit.osmocom.org/openbsc/commit/?id=faeec5cbf1c0430e64727c68f36b313dcdeaa709

commit faeec5cbf1c0430e64727c68f36b313dcdeaa709
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Mar 31 11:57:38 2013 +0200

    If requested TCH/H channel is not available, try assigning TCH/F
    
    If MNCC application requests a half rate channel, the channel might not be
    available, due to different cell configuration, so the full rate channel
    is used instead.

http://cgit.osmocom.org/openbsc/commit/?id=10772ea116e331e0bd63dc4c538e405412948941

commit 10772ea116e331e0bd63dc4c538e405412948941
Author: Andreas Eversberg <[email-masked]>
Date:   Wed Jan 22 08:57:18 2014 +0100

    Add check to tch_map(), if RTP sockets exist

http://cgit.osmocom.org/openbsc/commit/?id=1210e9f81a9f97608b264d40c6cdcdf24ef0cad6

commit 1210e9f81a9f97608b264d40c6cdcdf24ef0cad6
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Mar 31 11:50:35 2013 +0200

    Fix: If half rate channel is used, indicate hr when modifying channel mode

http://cgit.osmocom.org/openbsc/commit/?id=cac172562955f84e7c690a90fa5b424da5ebf083

commit cac172562955f84e7c690a90fa5b424da5ebf083
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Mar 31 11:44:34 2013 +0200

    Fix: If paging for half rate was requested, use hr, if supported by MS

http://cgit.osmocom.org/openbsc/commit/?id=d5bea29c5b6837c3655f16d638ce21127c1a9594

commit d5bea29c5b6837c3655f16d638ce21127c1a9594
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Jan 19 11:47:44 2014 +0100

    Add full AMR multirate IE support with VTY config for MS and BTS side

http://cgit.osmocom.org/openbsc/commit/?id=138b6583bffe94d99ef883cf63f686c566267ef2

commit 138b6583bffe94d99ef883cf63f686c566267ef2
Author: Andreas Eversberg <[email-masked]>
Date:   Thu Mar 14 09:41:18 2013 +0100

    Drop bad speech frames rather than forwarding them via RTP
    
    Some RTP endpoints may not check for bad frame indications, so a frame
    that is marked as bad may be still forwarded, which creates anoying noise.
    This patch drops these frames. It depends on the other RTP endpoint how
    dropped frames are handled. (insert silence, extrapolate speech...)

http://cgit.osmocom.org/openbsc/commit/?id=43cbefb7e56a66c693fda6c967b7007c493c243a

commit 43cbefb7e56a66c693fda6c967b7007c493c243a
Author: Andreas Eversberg <[email-masked]>
Date:   Mon Mar 11 08:20:48 2013 +0100

    Add check for non existing lchan at tch_frame_down()
    
    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.

http://cgit.osmocom.org/openbsc/commit/?id=0cee8c939b87bfb9c0513e44db7d30e6ad5ea51b

commit 0cee8c939b87bfb9c0513e44db7d30e6ad5ea51b
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Mar 10 11:50:19 2013 +0100

    Send RADIO LINK TIMEOUT value via OML attribute to BTS
    
    The same radio link timeout value is used for BTS and MS side.

http://cgit.osmocom.org/openbsc/commit/?id=5bb3813705ac854329f924b89054485884841d1c

commit 5bb3813705ac854329f924b89054485884841d1c
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Mar 10 11:49:35 2013 +0100

    Add option to set RADIO LINK TIMEOUT value via VTY

http://cgit.osmocom.org/openbsc/commit/?id=1118d8b5dc2d6a5b721048d60dfc2ea8148d3985

commit 1118d8b5dc2d6a5b721048d60dfc2ea8148d3985
Author: Andreas Eversberg <[email-masked]>
Date:   Fri Mar 16 08:14:23 2012 +0100

    Fixed problem of mute audio on some calls
    
    When reading from RTP socket, the first read() may fail right after
    connecting to remote socket. Subsequent read() will work as it should.
    
    If the remote socket does not open fast enough, the transmitted RTP
    payload can cause an ICMP (connection refused) packet reply. This causes
    the read to fail with errno=111. In all other error cases, the errno is
    logged at debug level. In all error cases, reading is not disabled.

http://cgit.osmocom.org/openbsc/commit/?id=c3d579ccaf3e52181c0fd1097cf578355ada9f6c

commit c3d579ccaf3e52181c0fd1097cf578355ada9f6c
Author: Andreas Eversberg <[email-masked]>
Date:   Thu Mar 8 11:08:37 2012 +0100

    Fixed (ipaccess BTS) delay/silence problems, if RTP stream jitters too much
    
    After OpenBSC stalled for some reason (e.g. CPU overload or database
    access) or after speech frames have been lost (MNCC application problems /
    hold/retrieve call), the timestamp and the sequence number of the RTP
    socket state must be corrected. The amount of incrmentation is calculated
    from the elapsed time. Not incrementing timestamp and sequence number would
    cause all frames to be dropped by ipaccess BTS, because the BTS expects
    frames with more recent timestamps.
    
    If speech frames are received too fast, they must be dropped. The timestamp
    and sequence number of the RTP socket state are not changed in this case.
    Incmenetating timestamp and sequence number would causes high delay at
    ipaccess BTS, because the BTS would queue the frames until the timestamp
    matches the current time.
    
    There is a simple test case:
    
    Make a call between two phones and check the delay. (When using LCR, make
    a call to the echo test.) The press CTRL+z to suspend OpenBSC process for
    a few seconds and enter "fg" to continue OpenBSC process. There shall be no
    change in the delay, even after repeating this test many times.

http://cgit.osmocom.org/openbsc/commit/?id=e39f6eba4c194c890ea2930013f4315d36472837

commit e39f6eba4c194c890ea2930013f4315d36472837
Author: Andreas Eversberg <[email-masked]>
Date:   Fri Feb 17 15:20:59 2012 +0100

    Allow dynamic RTP payload types between application and MNCC interface
    
    Since EFR/AMR/HR codecs use dynamic RTP payload, the payload type can
    be set. If it is set, the frame type must be set also, so OpenBSC
    knows what frame types are received via RTP.
    
    This modification only affects traffic beween application and MNCC
    interface, not the RTP traffic between OpenBSC and BTS.

http://cgit.osmocom.org/openbsc/commit/?id=5fea0d2c66d70e7c18b57c7d97fa16e3b752d010

commit 5fea0d2c66d70e7c18b57c7d97fa16e3b752d010
Author: Andreas Eversberg <[email-masked]>
Date:   Mon Jan 16 09:29:28 2012 +0100

    Add traffic forwarding via RTP to remote application
    
    Instead of forwarding traffic through MNCC interface, traffic can
    be forwarded to a given RTP peer directly. A special MNCC message
    is used to control the peer's destination. The traffic can still be
    forwarded through MNCC interface when this special MNCC message is
    not used.
    
    It also works with E1 based BTSs.
    
    In conjunction with LCR's "rtp-bridge" feature, the RTP traffic
    can be directly exchanged with a remote SIP endpoint, so that the
    traffic is not forwarded by LCR itself. This way the performance
    of handling traffic only depends on OpenBSC and the remote SIP
    endpoint. Also the traffic is exchanged with the SIP endpoint
    without transcoding, to have maximum performance.
    
    Increment MNCC version to 5.

http://cgit.osmocom.org/openbsc/commit/?id=cff7b114762b16b3d1be3988bd06428f2828b895

commit cff7b114762b16b3d1be3988bd06428f2828b895
Author: Andreas Eversberg <[email-masked]>
Date:   Fri Jan 20 20:33:37 2012 +0100

    Add handling of BFI (Bad Frame Indicatior) of received TRAU frames
    
    If a bad TRAU frame is received, it is forwarded to MNCC application
    as GSM_BAD_FRAME. The application can now handle the GAP of missing
    audio. (e.g. by extrapolation)
    
    If TRAU frames are forwarded via RTP, bad frames are dropped, but frame
    counter and timestamp of RTP sender state is incremented.

http://cgit.osmocom.org/openbsc/commit/?id=c5fa2c952e52e02ae83e9fc92f25b0af5dcb0ea7

commit c5fa2c952e52e02ae83e9fc92f25b0af5dcb0ea7
Author: Andreas Eversberg <[email-masked]>
Date:   Thu Mar 8 14:39:19 2012 +0100

    Add support for AMR frames to MNCC/RTP interface
    
    AMR rate is currently fixed to 5.9k.

http://cgit.osmocom.org/openbsc/commit/?id=70554aa565cb6fd8a2bd6d7aee4a94b951ef82e7

commit 70554aa565cb6fd8a2bd6d7aee4a94b951ef82e7
Author: Andreas Eversberg <[email-masked]>
Date:   Sun Feb 16 08:03:59 2014 +0100

    HACK: Make mgcp_test compile
    
    This is not a solution, just a workarround to get rid of llround()

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The OpenBSC GSM Base Station Controller (+MSC/HLR/SGSN)