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, jolly/32c3 has been created at f7637bc5155e14aabdcae1d3271a43ebbade6ef8 (commit)

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commit f7637bc5155e14aabdcae1d3271a43ebbade6ef8 Author: Andreas Eversberg jolly@eversberg.eu Date: Mon Jan 20 07:50:53 2014 +0100

Hack: Make Nokia Flexi-BTS work

http://cgit.osmocom.org/openbsc/commit/?id=4d255e2aaacda06f042abb14e2e0117a8...

commit 4d255e2aaacda06f042abb14e2e0117a8d2eb269 Author: Andreas Eversberg jolly@eversberg.eu Date: Sun Mar 31 12:03:12 2013 +0200

Allow handling of TCH/F and TCH/H frames from layer 4

http://cgit.osmocom.org/openbsc/commit/?id=de522c78ca8b4b225b4df880d7e28faa8...

commit de522c78ca8b4b225b4df880d7e28faa8134f3ad Author: Andreas Eversberg jolly@eversberg.eu Date: Sun Mar 31 12:01:25 2013 +0200

osmo-nitb support for codec negotiation

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=ab56d32c57c670084bc009757651db912...

commit ab56d32c57c670084bc009757651db9127d005b5 Author: Andreas Eversberg jolly@eversberg.eu 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.

http://cgit.osmocom.org/openbsc/commit/?id=60eefc6620cd709a7d3d26d5520916881...

commit 60eefc6620cd709a7d3d26d55209168812726156 Author: Andreas Eversberg jolly@eversberg.eu 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=df202379cb69eaa7ab12a9f568177436a...

commit df202379cb69eaa7ab12a9f568177436aae67129 Author: Andreas Eversberg jolly@eversberg.eu Date: Tue Mar 26 08:21:46 2013 +0100

Fixed AMR payload type selection and handling

http://cgit.osmocom.org/openbsc/commit/?id=b113b0bd35c9f964c1d513cde917d55ae...

commit b113b0bd35c9f964c1d513cde917d55aef9baf18 Author: Andreas Eversberg jolly@eversberg.eu 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 decoded, which creates anoying noise. This patch drops these frames. It depends on the SIP endpoint how these dropped frames are handled. (insert silence, extrapolate speech...)

http://cgit.osmocom.org/openbsc/commit/?id=d3ae9498a8d26fb1583f992dd64b734a6...

commit d3ae9498a8d26fb1583f992dd64b734a6f98018c Author: Andreas Eversberg jolly@eversberg.eu 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.

http://cgit.osmocom.org/openbsc/commit/?id=c42af135521031ba061a31bd256529972...

commit c42af135521031ba061a31bd256529972e078fbe Author: Andreas Eversberg jolly@eversberg.eu Date: Tue Feb 19 12:55:59 2013 +0100

Fix: TCHH/HR payloads are 15 bytes (ToC + 14 bytes of speech data)

http://cgit.osmocom.org/openbsc/commit/?id=fd0b0528be9cec102239e4ec935101e1b...

commit fd0b0528be9cec102239e4ec935101e1bf110f82 Author: Andreas Eversberg jolly@eversberg.eu Date: Thu Mar 8 14:39:19 2012 +0100

Finished support for all codecs (RTP bridge and MNCC interface)

The codes is not yet tested.

AMR rate is currently fixed to 5.9k.

http://cgit.osmocom.org/openbsc/commit/?id=7ef67185e692f2dff75d5c3952cc15dbd...

commit 7ef67185e692f2dff75d5c3952cc15dbd6586bb0 Author: Andreas Eversberg jolly@eversberg.eu Date: Thu Mar 8 11:08:37 2012 +0100

Fixed delay problems, if RTP stream jitters too much

The RTP stream is generated or forwarded by OpenBSC to nanoBTS. Due to switching of streams (hold/retrieve call), packet loss or even stalling of sender's process, the time stamp must be corrected. If outdated packets are received, they get dropped.

http://cgit.osmocom.org/openbsc/commit/?id=fbc3393bee719c8c4723cb089635ccb80...

commit fbc3393bee719c8c4723cb089635ccb804a12464 Author: Andreas Eversberg jolly@eversberg.eu 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=08f433e1a3e7d2c6e9d1244fb3d63266b...

commit 08f433e1a3e7d2c6e9d1244fb3d63266b69e2409 Author: Andreas Eversberg jolly@eversberg.eu Date: Fri Jan 20 20:33:37 2012 +0100

Adding 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_TCHF_BAD_FRAME. The application can now handle the GAP of missing audio. (e.g. with extrapolation)

If TRAU frames are forwarded via RTP, bad frames are dropped, but frame counter and timestamp of RTP sender state is increased.

http://cgit.osmocom.org/openbsc/commit/?id=00bfaf0f96d7899790d89f9d2605b40cf...

commit 00bfaf0f96d7899790d89f9d2605b40cf445bb03 Author: Andreas Eversberg jolly@eversberg.eu Date: Mon Jan 16 09:29:28 2012 +0100

Adding traffic forwarding via RTP to remote application

Instead of forwarding traffic through MNCC interface, traffic can now be forwarded to a given RTP destination. A special MNCC message is used for that. 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.

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