OpenBSC June 2017

openbsc@lists.osmocom.org

23 participants
42 discussions

libosmo-sccp: Build breakage on --disable-static
by Holger Freyther 03 Jun '17

03 Jun '17

Hi, I just looked into a build breakage when --disable-static is passed. The first failure is because of: libosmo_sigtran_la_LDFLAGS = -version-info $(LIBVERSION) \ -no-undefined \ -export-symbols-regex '^osmo_' The testcases use xua_* sua_ symbols but they are not exported. Shall we export them or shall we forbid --disable-static on libosmo-sccp? The second part is that I had preferred the sccp library to be static only to avoid having to do proper ABI checking/versioning. If we fix the above, maybe we have to fix this part too? holger

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Initial Suggestion | Merging osmo-sim-auth and PySIM and update on MNCC with OsmocomBB
by Gerard Lawrence Pinto 01 Jun '17

01 Jun '17

Greetings, Congrats on OsmoCon 2017! I have been working with OsmocomBB, osmo-sim-auth, mncc-python projects in Osmocom. I would like to contribute some code snippets with your acceptance. To begin with I have committed: I) MNCC with OsmocomBB - *https://github.com/GerardPinto/call-control-mncc-sap <https://github.com/GerardPinto/call-control-mncc-sap>* (Has both implementations in Python and C + Tested voice with gsm-fr with gsm audio file - works great). - Some of which can be committed to mncc-python? (In order to support MNCC with OsmocomBB too) Although, Harald told me it was built for OpenNITB - I'm not sure, the relevance of this contribution to this repo? Reason: 1. Solves some mailing list questions: Some questions on mailing list are pertaining sending voice from PC to phone and receive. 2. Useful for test cases: In the future, if 'gsm-efr' codec is implemented, we can use these snippets and send audio with gsm-efr sample, instead of connecting to LCR (Am I correct?) *Issues faced*: File: l23api.c | function: l1ctl_rx_traffic_req() 264 bits (33 bytes) of voice codec are interleaved into 456 bits. The DAC, ciphering/deciphering , RF etc. work on 114 bits at a time so, 114 x 4 = 456. So Layer23 sends 4 - 114 bits into the queue to Layer1 in function => l1ctl_rx_traffic_req *Code*: num = l1a_txq_msgb_count(&l1s.tx_queue[L1S_CHAN_TRAFFIC]) if num > 4 dropping traffic frame. (114 x 4) The code above indicates if queue has any (114 x 4) bits drop the incoming voice request. So I faced the issue of voice packets getting dropped, but that was just twice/thrice and then it did not give me any trouble (I'm not sure how it got fixed), I could hear *clear* voice gsm-fr on the called party cell phone. *My Question is*: If I happened to send voice packets 33 bytes really fast. There should be some kind of buffer rather than a check (Code) and drop frame? - OR this case can never happen? My apologies, if my understanding is incorrect, I just tried to reverse engg. from code and GSM codec available online! II) osmo-sim-auth with added other SIM related API's (still needs code clean up, SIM response check, code re-usable, README.md etc,) * https://github.com/GerardPinto/osmo-sim-auth <https://github.com/GerardPinto/osmo-sim-auth> * - I did check out PySIM (I think I can add some code to the existing repository, with your permission). Could you please let me know if my osmo-sim-auth repo? If it is correct to combine with PySIM, with the changes done below? Files modified: 1. SIM.py - Added more API's. 2. osmo-sim-auth.py - Options parser. Thanks, Gerard.

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OpenBSC June 2017