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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.
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Equipment like AudioCode appears to get upset when we use a
builtin type and then assign a name to it. Allow to completely
omit the name.
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This allows you to specify the osmux batch frame size. If zero, the
library uses the default value.
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via mgcp section from the configuration file.
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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.
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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.
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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
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This patch adds the get_net_downlink_format_cb() callback to provide
payload_type, subtype_name, and fmtp_extra suitable for use in a MGCP
response sent to the network. Per default, the BTS side values are
returned since these must be honoured by the net peer when sending
audio to the media gateway (unless transcoding is done).
Sponsored-by: On-Waves ehf
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This patch adds the callbacks rtp_processing_cb and
setup_rtp_processing_cb to mgcp_config to support arbitrary RTP
payload processing.
Sponsored-by: On-Waves ehf
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This patch adds the voice muxer. You can use this to batch RTP
traffic to reduce bandwidth comsuption. Basically, osmux transforms
RTP flows to a compact batch format, that is later on decompacted
to its original form. Port UDP/1984 is used for the muxer traffic
between osmo-bsc_nat and osmo-bsc_mgcp (in the BSC side). This
feature depends on libosmo-netif, which contains the osmux core
support.
Osmux is requested on-demand via the MGCP CRCX/MDCX messages (using
the vendor-specific extension X-Osmux: on) coming from the BSC-NAT,
so you can selectively enable osmux per BSC from one the bsc-nat.cfg
file, so we have a centralized point to enable/disable osmux.
First thing you need to do is to accept requests to use Osmux,
this can be done from VTY interface of osmo-bsc_nat and
osmo-bsc_mgcp by adding the following line:
mgcp
...
osmux on
osmux batch-factor 4
This just initializes the osmux engine. You still have to specify
what BSC uses osmux from osmo-bsc_nat configuration file:
...
bsc 1
osmux on
bsc 2
...
bsc 3
osmux on
In this case, bsc 1 and 3 should use osmux if possible, bsc 2 does
not have osmux enabled.
Thus, you can selectively enable osmux depending on the BSC, and
we have a centralized point for configuration from the bsc-nat to
enable osmux on demand, as suggested by Holger.
At this moment, this patch contains heavy debug logging for each
RTP packet that can be removed later to save cycles.
The RTP ssrc/seqnum/timestamp is randomly allocated for each MDCX that
is received to configure an endpoint.
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So far, a single dummy packet has been sent immediately after the
reception of a MDCX message. There is no dedicated keep alive
mechanism (it just worked because the audio from the MS has always
been forwarded to the NAT until the 'mgcp: Set output_enabled flags
based on the MGCP mode' patch).
This patch adds explicit, timer based keep alive handling that can be
enable per trunk. A VTY command 'rtp keep-alive' command is added for
configuration which can be used to set the interval in seconds, to
send a single packet after the reception of a CRCX/MDCX when RTP data
from the net is expected ('once'), or to disable the feature
completely ('no rtp keep-alive'). In 'send-recv' connections, only
the initial packet is sent if enabled (even when an interval has been
configured). The default is 'once'.
Note that this removes the mgcp_change_cb() from mgcp_main.c.
Sponsored-by: On-Waves ehf
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Currently, all timestamps are force to SeqNo*d + C which is more than
required by the nanoBTS which seems to be sensitive to alignment
errors only (dTS != k*d, d = ptime * rate = 160).
This patch replaces the force_constant_timing feature by a
force_aligned_timing feature. The timestamp offset will only be
changed (and timestamp errors counted) when the alignment does not
match to the raster based on ptime (default 20ms).
The VTY interface does not change.
Sponsored-by: On-Waves ehf
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Currently the SDP 'ptime' media attribute is never set in generated
MGCP responses.
This patch optionally includes the 'ptime' attribute if
packet_duration_ms is != 0. This behaviour can be enabled/disabled
by using the VTY command "sdp audio-payload send-ptime" (enabled by
default).
Sponsored-by: On-Waves ehf
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This adds datastructures and a VTY frontend to configure the
different type of RTP header patching: SSRC and timestamp.
Note that timestamp patching is not yet implemented.
Sponsored-by: On-Waves ehf
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This enum indicates if the mgcp is running on the BSC or the BSC-NAT.
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We want to send a TRAP with the MGCP statistics from the NAT and
the connected BSC. The BSC endpoint can be either released because
of a DLCX from the MGCP CallAgent or the SCCP Connection release on
the A-link.
This is why we need to queue the statistics when the deleting the
endpoint on the BSC. The processing is continued once the response
arrives. This code assumes that the response of the DLCX will be sent
by the remote side. The current amount of outstanding responses can be
seen on the VTY. This assumption is based on the fact that the BSC has
already responded to the CRCX and maybe to the MDCX.
The MGCP RFC is bended to prefix the transaction identifier with "nat-"
to easily detect the response and hand it to the handler. This will
then parse the response and generate the TRAP. The current version is
v1. We assume that the transaction space is big enough and we will
not re-assign the transaction identifier too early.
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In case of AMR one can specify the available codecs out-of-band. Allow
to configure this line statically in the configuration file.
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MGCP is used over UDP and a response might be lost. The MGCP RFC
asks for keeping a list of responses and then using the previous
response to answer a duplicate request. I tried to conserve memory
and just wanted to remember the last transaction identifier and
result-code and re-generate the result from that. This made the
code look bad and this is why the entire response will now be stored.
It sadly increases the memory usage but can not be avoided at this
time.
Remove the msg->l3h pointer for the RQNT callback as strtok has
modified the content of it.
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This is required for remembering the last result in a central place
and is required to properly implement detection of re-transmissions.
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Introduce a callback for the request and forward the signalrequest
to the callback. This is not a full implementation of MGCP RQNT.
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Follow the MGCP specification and send the collected statistics
at the end of a call. Right now this does not include jitter, packet
loss and delay.
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The ip.access nanoBTS appears to send quite broken NTP timestamps in
the RTCP messages might confuse equipment that uses the sender report
of the BTS. Make it easy to experiment by adding an option to drop RTCP.
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Implement the RSIP spec extension to work on the specified
trunk instead of hardcoding it to the virtual trunk.
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Summary of changes:
s/struct write_queue/struct osmo_wqueue/g
s/write_queue_init/osmo_wqueue_init/g
s/write_queue_clear/osmo_wqueue_clear/g
s/write_queue_enqueue/osmo_wqueue_enqueue/g
s/write_queue_bfd_cb/osmo_wqueue_bfd_cb/g
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libosmogsm is a new library that is distributed in the libosmocore.
Now, openbsc depends on it. This patch gets openbsc with this
change.
This patch also rewrites all include path to the new
osmocom/[gsm|core]
Signed-off-by: Pablo Neira Ayuso <pablo@gnumonks.org>
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A trunk will always have 32 endpoints to be used and we allow
a sparse allocation of endpoints.
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We want to support real trunks in the MGCP code and we need to
have some better book keeping for those. Move the code around.
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Switch to a \r\n at the end of the query. Both \n and \r\n
are valid according to the spec but this makes us match
responses of other equipment.
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The reason for this is quite simple: We want to make sure anyone
running a customized version of OpenBSC to operate a network will
have to release all custom modifiations to the source code.
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The transcoder RESET is using the same extensions to reset all
endpoints on a remote site. This makes sure that all allocations
can be made in a properly configured network.
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The code assumes a static mapping from endpoint to RTP port,
be able to configure the start of such a mapping.
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Move the bfd from a static var into the mgcp_config.
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The timeslot one is blocked and should not be used, replace the
code with a warning and watch out for it. Tis is most likely due
the uncovered in the previous commit due the wrong TLV definition.
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We really have 32 channels per multiplex... so use the right
number... or at least it seems we do have 32.
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Use the right source port when sending the message.
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Simplify the code by onlt allowing one way to allocate
a socket.
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We plan to have two different ports for the network and for the
BTS to avoid detecting the BTS and to dynamically allocate the
port to have old data not go to a new socket.
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