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This is causing a regression with AMR audio. The nanoBTS
is sending a MDCX NACK with Cause 0x52. Reverting this
commit makes it work again.
This reverts commit b54dda4cefe89665d98074cf3767858fa6ecdb6e.
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For GSM V1 FR, the payload type is fixed to 3 in the RFC.
But for the other codecs, the payload type is dynamically assigned
between 96 and 127. Here, we use a static mapping internal to OpenBSC.
This patch is needed to make a rather old 139 unit (with sw version
120a002_v149b42d0) work with something else than FR codec. I also tested
this patch on a newer 139 (with sw version 120a352_v267b22d0) to make
sure it didn't add a regression. More testing with newer EDGE units
should be done by whoever has some of theses.
Signed-off-by: Sylvain Munaut <tnt@246tNt.com>
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This library is intended to collect all generic/common funcitionality
of all Osmocom.org projects, including OpenBSC but also OsmocomBB
The library currently includes the following modules:
bitvec, comp128, gsm_utils, msgb, select, signal, statistics, talloc, timer,
tlv_parse, linuxlist
msgb allocation error debugging had to be temporarily disabled as it depends on
'debug.c' functionality which at the moment remains in OpenBSC
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During handover, we will not send RTP frames for quite some time. However,
the way the rtp_send code is structured, it will increment the timestamp
with a fixed amount every time we send a frame, independent how much wallclock
time has actually passed.
This code is a hack to update the sequence number and timestamp in case it
seems to be wrong. It makes handover much more reliable.
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Instead of passing TRAU frames down the MNCC API to the call control
application like MNCC, we now decode the TRAU frame into the actual codec
frame. We do the same with the RTP packets in case of ip.access and
thus have a unified format of passing codec data from the BTS to
an application, independent of the BTS type.
This is only implemented for V1 full-rate at the moment, and needs
to be fixed.
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The rtp_proxy.[ch] code is intended to be used as a transparent
RTP/RTCP proxy, relaying the media streams from one ip.access BTS
to another. In an 'ideal' network, this is obviously not needed,
since the BTS's can send those streams directly between each other.
However, for debugging, 'lawful interception', transcoding or interfacing
a TRAU/E1 based BTS, we actually need to process those RTP streams
ourselves.
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