Use the existing ulaw encode/decode to support PCMU as well.
The MERA VoIP switch has some severe issues with the GSM codec
and it appears easier to enable transcoding for it.
The mera switch doesn't appear to cope with codec change
between a SIP 180 trying and the 200 ok connection result.
Inserting the codec is touching too many places. Ideally we
should have the transcoding function as pointer in the struct
as well but the arguments differ.. so it is not a direct way
forward.
Currently the src_codec const variable is set to &src_end->codec
before src_end is checked against NULL. Since the assigment is just
an address operation and the memory where it points to is only
accessed after the NULL check, this does not harm technically.
Nevertheless this is potential source for errors if that code is
changed.
This commit moves the definition below the NULL check. This does not
comply with the coding style, but it cannot be split into definition
and a later assignment due to the const qualifier.
Sponsored-by: On-Waves ehf
We might have compiled transcoding into the MGW but
we don't want to enable it for a given user. Add a new
switch that should allow that.
I had manually tested the allow-transcoding/no allow
VTY interface for the primary interface and a new trunk
using show running-config.
It is unlikely that GSM, gsm and GsM refer to different codecs.
The mera mvts does send the audio codecs in lower case even if
RFC 3551 has them in upper case (but copy and paste is sometimes
too hard).
In case we get offered G729 and G711 we might have selected
G729 as the audio codec. The first packet we receive might be
G711 though. In that case we will need to change. But only if
we have a matching alternate codec payload_type. E.g. in the
case of comfort noise we will receive the PT=11 and we don't
want to change.
We might be offered multiple codecs by the remote and need to
switch between them once we receive data. Do this by moving it
to a struct so we can separate between proposed and current
codec. In SDP we can have multiple codecs but a global ptime.
The current code doesn't separate that clearly instead we write
it to the main codec.
Use the rtp_hdr structure. The basic alignment issue remains
and I need to merge/cherry-pick Jacob's getters for the ts,
sequence number and other attributes.
In case the sender didn't send a couple of frames we will have
a time gap that is bigger than the accepted delta. Add a new
testcase for this and update the next_time.
Transcoding from GSM to PCMA can lead to the MGCP MGW sending
two PCMA packages with the same sequence number and timestamp.
Once with the encoded audio and once completely empty.
This is because "state->dst_packet_duration" is 0 in most cases
(unless a ptime is forced) and we attempt to encode audio even
if there are not enough samples. The encode_audio return will
return 0 in that case which is not trated as an error by the
mgcp network code.
Handle rc == 0 specially and document the semantic.
The sequence number was read from the wrong place and then
the wrong byte order conversion routine was used so we ended
up wirting 0x00, 0x00 into the patched sequence number. Add
a testcase for that.
The GSM handle was never released. This was found using valgrind
and the leak check.
==14933== 752 bytes in 1 blocks are definitely lost in loss record 15 of 19
==14933== at 0x4028B4C: malloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so)
==14933== by 0x4130201: gsm_create (in /usr/lib/i386-linux-gnu/libgsm.so.1.0.12)
==14933== by 0x80517AE: mgcp_transcoding_setup (mgcp_transcode.c:199)
==14933== by 0x8049691: given_configured_endpoint.isra.1 (mgcp_transcoding_test.c:198)
==14933== by 0x8049C11: test_transcode_result (mgcp_transcoding_test.c:328)
==14933== by 0x8049418: main (mgcp_transcoding_test.c:582)
This patch moves the files relevant to transcoding from
src/osmo-bsc_mgcp to src/libmgcp and src/include/openbsc. Makefiles
and include directives are being updated accordingly.
Sponsored-by: On-Waves ehf
Holger Hans Peter Freyther