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Most of the "fixes" have nothing to do with gcc-4.9.2 but are a
question of ABI/Architecture (e.g. x86 vs. AMD64). Revert these
for now.
This partially reverts commit 7b1d25a11e44bbc1cb0d2acd9f1a3d4a16ec7c90.
abis_test.c: In function ‘test_simple_sw_config’:
abis_test.c:68:2: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 2 has type ‘int’ [-Wformat=]
printf("Start: %ld len: %zu\n", descr[0].start - simple_config, descr[0].len);
^
abis_test.c: In function ‘test_dual_sw_config’:
abis_test.c:111:2: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 2 has type ‘int’ [-Wformat=]
printf("Start: %ld len: %zu\n", descr[0].start - dual_config, descr[0].len);
^
abis_test.c:115:2: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 2 has type ‘int’ [-Wformat=]
printf("Start: %ld len: %zu\n", descr[1].start - dual_config, descr[1].len);
^
abis_test.c: In function ‘test_sw_selection’:
abis_test.c:132:2: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 2 has type ‘int’ [-Wformat=]
printf("Start: %ld len: %zu\n", descr[0].start - load_config, descr[0].len);
^
abis_test.c:136:2: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 2 has type ‘int’ [-Wformat=]
printf("Start: %ld len: %zu\n", descr[1].start - load_config, descr[1].len);
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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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For jitter, transit and packet loss we should count the data
that arrived and not the data we send towards the remote. This
is changing the jitter timings to what they were before the
re-factoring.
For forced timing we might willingly add jumps in the sequence
number but for jitter and packet loss we are more interested
in the data that traveled through the wire/air.
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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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This allows us to know what number of messages and bytes has been
received per active osmux endpoint.
Note that an Osmux message is composed of several chunks. Each chunk
contains an osmux header plus several voice data frames.
P: PS=385, OS=11188, PR=195, OR=5655, PL=0, JI=49
X-Osmo-CP: EC TIS=0, TOS=0, TIR=0, TOR=0
X-Osmux-ST: CR=51, BR=3129
The new 'X-Osmux-ST:' notifies the received chunks and bytes.
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Use struct mgcp_rtp_end statistics to account the RTP messages
that has been extracted from the osmux batch and transmitted.
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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.
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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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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.
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The osmux header uses a counter of 3 bits, so you can put up to
8 message in it.
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Just like other options do, to avoid polluting the configuration file
with unused options if osmux is disabled.
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The library allows to indicate zero as batch size if you want to use
the default size, however openbsc saves 'osmux batch-size 0' which is
not good as input.
Use OSMUX_BATCH_DEFAULT_MAX to explicitly initialize the batch size
from mgcp_parse_config().
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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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** CID 1232804: Unchecked return value (CHECKED_RETURN)
/src/libmgcp/mgcp_protocol.c: 888 in mgcp_parse_osmux_cid()
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The callback is responsible for releasing the batch message that
libosmo-netif builds.
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mgcp_send() needs some initialized address when printing a log message.
Nothing really serious but let's calm down valgrind.
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So we can easily identify in the log message what refers to
libosmo-netif and what to libmgcp.
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It may print a debug line every 20ms, so disable this. We can still
compile this extra spamming debug from the libosmo-netif library.
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Instead of the hardcoded OSMUX_PORT.
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via mgcp section from the configuration file.
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This patch adds a missing goto err. While at it, reword log message.
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Not very useful:
<000b> osmux.c:163 Osmux uses CID 1 from endpoint=7 (active=1)
Get rid of it.
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Use osmux_xfrm_input_fini() to release the internal state of the osmux
input handle.
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This patch includes several osmux fixes that are interdependent:
1) This adds Osmux circuit ID, this is allocated from the bsc-nat. This
announces the circuit ID in the CRCX MGCP message. This aims to resolve
the lack of uniqueness due to the use of endp->ci, which is local to
the bsc. This ID is notified via X-Osmux: NUM where NUM is the osmux
circuit ID.
2) The dummy load routines are now used to setup osmux both in bsc and
bsc-nat to resolve source port NAT issues as suggested by Holger. The
source port that is used from the bsc is not known until the first
voice message is sent to the bsc-nat, therefore enabling osmux from
the MGCP plane breaks when a different source port is used.
3) Add refcnt to struct osmux_handle, several endpoints can be using the
same input RTP osmux handle to perform the batching. Remove it from the
osmux handle list once nobody is using it anymore to clean it up.
4) Add a simple Osmux state-machine with three states. The initial state
is disabled, then if the bsc-nat requests Osmux, both sides enters
activating. The final enabled state is reached once the bsc-nat sees
the dummy load message that tells what source port is used by the bsc.
5) The osmux input handle (which transforms RTP messages to one Osmux batch)
is now permanently attached to the endpoint when Osmux is set up from the
dummy load path, so we skip a lookup for each message. This simplifies
osmux_xfrm_to_osmux().
After this patch, the workflow to setup Osmux is the following:
bsc bsc-nat
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|<------ CRCX ----------|
| X-Osmux: 3 | (where 3 is the Osmux circuit ID
| | that the bsc-nat has allocated)
|------- resp --------->|
| X-Osmux: 3 | (the bsc confirm that it can
| | use Osmux).
. .
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setup osmux |----- dummy load ----->| setup osmux
| Osmux CID: 3 |
In two steps:
1st) Allocate the Osmux Circuit ID (CID): The bsc-nat allocates an unique
Osmux CID that is notified to the bsc through the 'X-Osmux:' extension.
The bsc-nat annotates this circuit ID in the endpoint object. The bsc
replies back with the 'X-Osmux:' to confirm that it agrees to use Osmux.
If the bsc doesn't want to use Osmux, it doesn't include the extension
so the bsc-nat knows that it has to use to RTP.
2nd) The dummy load is used to convey the Osmux CID. This needs to happen
at this stage since the bsc-nat needs to know what source port the bsc
uses to get this working since the bsc may use a different source
port due to NAT. Unfortunately, this can't be done from the MGCP signal
plane since the real source port is not known that the bsc uses is not
known.
This patch also reverts the MDCX handling until it is clear that we need
this special handling for this case.
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In the bsc-nat side, the osmux socket initialization can be done from
the vty. This ensure that the osmux socket is available by the time the
bsc-nt receives the dummy load that confirms that the osmux flow has
been set up.
This change is required by the follow up patch. This change ensures that
the Osmux socket in the bsc-nat is already in place by the time this
receives the dummy load.
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This is a cleanup to allow the reuse of the new functions
osmux_handle_find_get() and osmux_handle_alloc().
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Just a cleanup, wrap around the osmux state information in a struct.
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Avoid creating a bogus state that will never go away.
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For our usecase several different systems might be behind the
same firewall so we need to distinguish the remote by more than
the IPv4 address.
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Set the remote port for the endpoint. Somehow this needs to propagate
all the way to the handle.
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We need to discover the remote port as we are likely behind a NAT.
Right now the NAT code will just send to port 1984 on the BSC but
this might not arrive at the BSC. Include the CI (in the future we
need to include the endpoint address or send the dummy to the net
port). This is just an interim solution.
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The CI is a MGCP value that is counted from 0 upwards. The code
is comparing a uint8_t with a uint32_t. This will only work for
up to UINT8_MAX calls and then will silently break. The code should
probably work with the endpoint number and not the CI. For now
truncate things and hope things work.
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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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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.
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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 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.
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When going from a ptime of 10 to 20 a lot of alignment errors
are reported. In fact the alignment check should be done before
and after the transcoding. As this is not possible right now
only do it _after_ the patching.
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For the unit tests we need to look at the internal state.
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When the sample buffer is being dropped write a warning so we can
see how often this event occurs.
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