![]() 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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openbsc.pc.in | ||
osmoappdesc.py |
README
About OpenBSC ============= OpenBSC is a minimalistic implementation of the GSM Network, with particular emphasis on the functionality typically provided by the BSC, MSC, HLR, VLR and SMSC. Its currently supported interfaces towards the BTS are: * Classic A-bis over E1 using a mISDN based E1 interface. In other words, you can connect existing GSM Base Transceiver Station (BTS) through E1 to OpenBSC. So far, we have only tested the Siemens BS-11 Test reports with other BTS are much appreciated! * A-bis over IP as used by the ip.access nanoBTS product family You can find the project documentation at http://openbsc.gnumonks.org/ This project is still in its early days, and there are lots of areas where it doesn't behave as per GSM spec. Harald Welte <laforge@gnumonks.org> libosmocore =========== Please note that as of March 2010, OpenBSC has a dependency to a library called "libosmocore". You can obtain that library from git://git.osmocom.org/libosmocore.git