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.
Jacob pointed out that "free_endp" refers to the memory of
the endpoint being freed. What we want is actually a way to
release an endpoint (and the resource it allocated) or in
the case of the testcase/testapp initialize the data structure
correctly. Introduce two names for that.
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
The current transcoder implemenation always does a 1:1 recoding
concerning the duration of a packet. So RTP timestamps and sequence
numbers are not modified.
This is not sufficient in some cases, e.g. when the BTS does only
allow for a single fixed ptime.
This patch decouples encoding from decoding and moves the decoded
samples to the state structure so that samples can be combined or
drain according to the packaging of incoming and outgoing packets.
This patch incorporates parts of Holger's experimental fixes in
0e669e05^..9eba68f9.
Ticket: OW#1111
Sponsored-by: On-Waves ehf
This adds a --frame-size option to read payload binary files with a
fixed frame size directly. The file must not contain RTP headers.
In addition '--rate' and '--duration' can be used to configure the
timing.
Sponsored-by: On-Waves ehf
There is the wrong record field selection being used to extract the
default value. It returns the tuple offset instead of the value.
This patch fixes this.
Sponsored-by: On-Waves ehf
This tool provides the following features:
- Output formats: state, C arrays
- Optionally take RTP payload from existing state files
- Generate streams with RTP timestamp jumps and/or delays
- Set/change SSRC or payload type
Requires erlang to be installed.
Example:
Generate 300 packets, set playout time offset to 1s, set
RTP timestamp offset to 8000 (1s), generate another 100
packets, the RTP payload is copied from rtp.state:
./gen_rtp_header.erl --type=98 --file=rtp.state --
0 300 0 --delay=1.0 100 8000
Sponsored-by: On-Waves ehf
This patch adds optional parameters to pass the state file, the
destination address (default 127.0.0.1), the destination port
(default 4000), the source port (default 0). So it is called as
follows:
gst rtp_replay.st -a [FILE [HOST [SOURCEPORT [DESTPORT]]]]
In addition, nonexistant FILEs are no longer created but opened
read-only instead.
Sponsored-by: On-Waves ehf
The test scripts warn about missing documentation, untested configs,
check common errors, and stub out testing individual VTY commands.
The scripts have been moved to the another osmocom repository,
python/osmo-python-tests
The features were requested by zecke.
Use the Smalltalk SIP implementation to create a call
and once the call has been established start the replay
using the commoncode. No patching of RTP occurs yet.
Update/Move/Create example configuration files for NiTB, BSC,
MGCP, NAT and the GbProxy. Create a script that starts, generates
the vty reference and terminates the application.
This can be used to throw the data into GNUplot. It collects
the time (from the start of the trace), the buffer data in kbyte
and the number of buffered PDUs. It is assuming that no PDU
is delivered toward the target.
This is creating 1000 subscribers and 30 SMS each. The SMS
itself is badly formatted (not a valid 7bit encoding) but
it should be enough for a stress test.
Split the trace based on SCCP Connections, in case an error is
detected, the trace will be stored instead of being deleted and
the full log can be viewed afterwards.
Right now only LU Rejects are recognized, but this can be extended
to other things as well.
Henning Heinold
Daniel Willmann
Harald Welte