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Holger reports that the bitmap that accounts for available Osmux circuit
IDs is limited to 128, when the maximum number of circuit IDs are
determined by the uint8_t field in the header (ie. 256 circuits).
[hfreyther: Update the testcase now that we have more ids to allocate]
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sizeof(uint8_t) == 1 and there is no need to create an array
with 16 bytes and then only use the first two of them. This
means the CID range is from 0 to 127 and we should be able
to extend this to 256 by changing the array size to 32. Update
the testcase now that we can have more than 16 calls with Osmux.
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* Test that one can get an id
* That they are assigned predicatble right now
* That returning them will make the number of used ones go down
* That allocating more will fail
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We want the port and IPv4 address to be extracted but don't
expect any change in audio codecs.
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The NAT sends an incomplete SDP file for the purpose of informing
the BSC about the remote IP/PORT early. The case of an incomplete
SDP file was not considered. Check if there is a codec and if not
skip it.
TODO: We need to have a better end-point life cycle test.
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We have a lot of legacy that I am afraid to break. We have
everything in place to make a good codec selection (e.g. if
we can avoid transcoding, pick the one with best quality or
the lowest speed). Right now I have a specific case where
from all options I want to pick GSM. Guard the codec compat
check behind the disallow transcoding option to make sure
to not break legacy application.
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The session name must be present in a SDP file. The RFC proposes
to use a space for it but the other equipment is using the dash
so I have picked that as well.
RFC 4566:
The "s=" field is the textual session name. There MUST be one and
only one "s=" field per session description. The "s=" field MUST NOT
be empty and SHOULD contain ISO 10646 characters (but see also the
"a=charset" attribute). If a session has no meaningful name, the
alue "s= " SHOULD be used (i.e., a single space as the session
name).
Fixes: RT#2196
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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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The Annex A code has a probation period but we don't have it. When
starting with seq_no==0 do not assume that the sequence numbers
have wrapped. Do it by moving the entire checking code into the
else.
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mgcp_patch_and_count has grown due supporting linearizing timestamps,
ssrc and other things for equipment like the ip.access nanoBTS. Fight
back and move the Annex A code into a dedicated method.
The result is updated as we now count after all the patching and for
the Annex A code no change in SSRC can be detected.
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In case of some RTP proxy from time to time we are offered both
G729 and G711 but only one of them will work. I intend to adjust
the codec at runtime in case we receive the wrong codec.
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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.
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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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Add tests setting the fmtp_extra field to check the response
generation. This triggers a bug found by Coverity.
Addresses: Coverity CID 1220873
Sponsored-by: On-Waves ehf
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Currently, if there is no SDP data in the MGCP message received from
the net, the fields containing audio encoding information are not set
in net_end. So in recvonly mode transcoding would not be set up
correctly.
This patch changes the implementation of the code handling CRCX and
MDCX to use the codec signalled in the MGCP local connection options
(field 'a:') if there isn't any SDP data. This is only halfway
negotiation, because the codec is used blindly and not matched
against the supported ones.
Sponsored-by: On-Waves ehf
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In ae1997248ccb4fba1394267d3051082dfd85448a the handwritten tokenizer
was replaced with strtok_r. As part of this change the structural
checking of MGCP parameters was stopped. This means that a code like
"line + 3" might access beyond the first NUL and be possibly behind
the msgb. Manually add size checking again. Manually jumping to the
error label is not possible anymore as it has been removed. The result
is that invalid lines will be skipped. This is matching the general
approach by the IETF RFCs to be permissive in data being received.
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With the current multiplication we might end up with 19999 as
time on i386. When we round it ends up as 20000 on i386 and
should work the same on AMD64.
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On AMD64 we had a difference in the test result most likely due
the bigger size of integers.
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Should fix on AMD64:
mgcp_test.c: In function ‘sendto’:
mgcp_test.c:318:10: warning: format ‘%d’ expects argument of type ‘int’, but argument 5 has type ‘size_t’ [-Wformat]
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Currently the stats (jitter, transit) cannot be checked properly,
since they depend on the wallclock time.
This patch fakes clock_gettime (CLOCK_MONOTONIC) to reflect the
scheduling time of the RTP packets. In addition, the RTP statistical
value are written to stdout.
A RTP test case with a SSRC change along with a reference time delay
has been added.
Sponsored-by: On-Waves ehf
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Currently the conn_mode and the output_enabled flags are printed to
stdout.
This patch modifies this to print the output_enabled flags to stderr
instead. The bits in conn_mode are shown as RECV, SEND, and LOOP.
This does not reduce the significance of the test, since there is an
assertion already that verifies the values of the output_enabled
flags with respect to the conn_mode.
Sponsored-by: On-Waves ehf
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Currently, the conn_mode field is reset after it has been checked.
This patch disables this behaviour and only adds a mark (bit) to
detect modifications.
Sponsored-by: On-Waves ehf
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This patch changes implementation and the mgcp_connection_mode enum
in a way that net_end.output_enabled (bts_end.output_enabled) flag
always matches the MGCP_CONN_SEND_ONLY (MGCP_CONN_RECV_ONLY) bit of
conn_mode.
Based on this, the conn_mode bits are then used instead of the
output_enabled fields within mgcp_protocol.c.
Sponsored-by: On-Waves ehf
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This patch extends the test_messages test to show the number of dummy
packets that has been passed to sendto(). This way, dummy packets are
even being counted when sendto() itself failes (e.g. due to the fd
being -1).
Sponsored-by: On-Waves ehf
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Currently these value may leak into the next test.
This patch will reset them after each message has been processed.
Sponsored-by: On-Waves ehf
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This modifies the test MGCP messages to include M:recvonly (CRCX),
M:sendrecv (MDCX) and M:sendonly (MDCX) outputs the resulting
output_enabled flags and the conn_mode after processing the message.
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 counter and output timestamp values are written out for
each packet. This makes it difficult to see in the diffs what has
been changed significantly.
This patch changes this by showing differences for those
values. The absolute input values are also shown now. In addition,
the sequence numbers (the difference for the output value) are
written, too.
Sponsored-by: On-Waves ehf
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This adds two test cases:
1. Packet repetition (dSeq=0, dTS=0)
2. Broken seqNo (dSeq=0, dTS=160)
The second had been already present in the test cases, but it was a
mere copy&paste mistake which turned out to be rather helpful. This
patch therefore turns it into a documented test case.
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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Currently the local connection options have been stored as a string.
This patch replaces this string by a struct (that still contains a
string) along with the parsed fields (only the packetization period
at the moment).
It also re-adds the calls to set_local_cx_options() to the
handle_create_con() and handle_modify_con() functions. Except for
the test program this has no side effects, since the LCO values
aren't used yet.
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This also adds additional MDCX tests (based on MDCX4) to test the
analysis of different combinations of 'p' and 'ptime' fields.
Sponsored-by: On-Waves ehf
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The hardening flags of debian have highlighted this sprintf mis-usage
in the testcase. Address it.
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The ssrc has been used without respect to proper byte ordering in
mgcp_patch_and_count(). This only affected log messages.
This patch introduces a new variable 'ssrc' that takes the value of
the SSRC in proper byte order.
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 patch extends the existing RTP error check test by adding a
check for timestamp errors after SSRC changes and a check for a
segno delta of 2 (with a timestamp delta of 320).
To test SSRC patching too, a corresponding line will be written on
each SSRC change that has been detected in the output stream.
In addition there is now support for selectively enabling/disabling
SSRC and timestamp patching. The RTP test sequence is repeated for
all combinations thereof.
Sponsored-by: On-Waves ehf
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So far the SDP part of the CRCX message has been ignored by the MGW.
This patch adds SDP parsing for this case, eventually updating the
net end's payload type and connection parameters.
Sponsored-by: On-Waves ehf
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These tests mainly check whether the SDP parsing works properly by
looking at the payload type detected.
Sponsored-by: On-Waves ehf
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This patch add the for_each_line macro based on a strline_r()
function (similar to strtok_r()), that is also part of this patch.
This strline_r() function is tolerant with respect to line endings,
it supports CR-only, CRLF, and LF-only and any combinations thereof
(note that a CRLF is always detected as a single line break).
Similar to for_each_non_empty_line (the former for_each_line) where
the 'save' pointer needed to be initialised by a call to strtok_r(),
the new for_each_line macro expects, that the 'save' pointer has been
initialised by a call to strline_r(). Also note, that
for_each_line/strline_r and for_each_non_empty_line/strtok_r may use
the 'save' pointer differently, so calls to them can not be mixed.
Sponsored-by: On-Waves ehf
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The current implementation increments the seqno but does not increment
the RTP timestamp, leading to two identical timestamps following one
after the other.
This patch fixes this by adding the computed tsdelta when the offset
is calulated. In the unlikely case, that a tsdelta hasn't been
computed yet when the SSRC changes, a tsdelta is computed based on
the RTP rate and a RTP packet duration of 20ms (one speech frame per
channel and packet). If the RTP rate is not known, a rate of 8000 is
assumed.
Note that this approach presumes, that the per RTP packet duration
(in samples) is the same for the last two packets of the stream being
replaced (the first one).
Sponsored-by: On-Waves ehf
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This patch adds a test case to check, whether RTP timestamps are
generated properly after SSRC changes and whether the error counters
work properly.
Sponsored-by: On-Waves ehf
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This patch modifies the patch_and_count() function to check for RTP
timestamp inconsistencies. It basically checks, whether dTS/dSeqNo
remains constant. If this fails, the corresponding counter is
incremented. There are four counter for this: Incoming and outgoing,
each for streams from the BTS and the net.
Note that this approach presumes, that the per RTP packet duration
(in samples) remains the same throughout the entire stream. Changing
the number of speech frames per channel and packet will be detected
as error.
In addition, the VTY command 'show mgcp' is extended by an optional
'stats' to show the counter values, too.
Ticket: OW#964
Sponsored-by: On-Waves ehf
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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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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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The second CRCX would be mostly ignored and be handled by the
retransmission layer of MGCP.
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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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Calculate the expected packages and packet loss as of RFC 3550.
The values should be clamped but our packet loss counter is 32
bits and not 24 and we should clamp at other values but I am
waiting for some issues first before dealing with that.
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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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