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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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