hi, here is the patch to support audio exchange between application interface and nanoBTS. the patch changes the audio format from TRAU frame format to RTP payload format on the application interface. also the BS11 TRAU frame is converted to/from RTP payload format (inside trau_mux.c), so an application does not have to care about what BTS is used. here is the format as defined by RFC: (just fyi) first byte has the magic value of 0xd0 plus four upper bits of the LARc[0] element as lower bits. second byte has the lower two bits of the LARc[0] element as upper bits and the upper bits of LARc[1] element as lower bits. third byte has the lower four bits of LARc[1] element as upper bits and so on... this format can directly be transcoded with the open source lossy gsm codec. the rtp_proxy.c now parses RTP frames and generates them. for easier code, i just mirrored the received RTCP frames back to the nanoBTS, mangled of yourse. (oops, maybe i need to change some synchronization source identifiers.) i think there is no problem with that right now. because RFC requires special synchronization source (ssrc) generation, i added md5 code and the (funny) random number generator. (see rtp_proxy.c random32()) as soon as the patch is applied, i will also check in linux-call-router modification for the new audio frame format. note that this code is just quickly tested, it works with both BS11 and nanoBTS and should not crash on corrupt or extended RTP data. the delay of audio data sent to nanoBTS was > 0.5 seconds. i think this is too much. maybe it is because i seamlessly increment the timestamp value in the RTP frame, even if the application dropps (delays) frames. regards, andreas