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Introduce the bad_seq handling that is dealing with a very
long sequence number jump. The only missing part is the
probation handling that is of not that much interest right
now.
Change the initialization sequence, in case of a new SSRC
re-set the jitter and transit time calculation to an initial
value. The sender might be a different system that takes a
different path so the average jitter does not make that
munch sense.
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The initialization would put max_seq to seq - 1 while init_seq
of the annex does not. Move the max_seq assignment into all the
branches.
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Somehow we end up with calculations of 65536 for the packet loss
that is RTP_SEQ_MOD. So there might have been a sequence overflow
but no packets to count against it. I have no idea how this can
happen and could not reproduce it. But when we change the SSRC
we should re-initialize the state. Separate the total packets and
octets from the packets we count per SSRC.
Related: OW#1489
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* Implement the rtp-bridge mode for MNCC
* Audio doesn't flow through the NITB at all
* It only works with IPv4 BTSes right now
* We need to select an audio codec way too early
* No tandem free operation
* Early assignment always equals TCH/F
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In case of the RTP bridge mode we need to select the codec
ourselves. Rely on the same (incomplete) codec selection that
can be done using the mncc-int configuration node. This might
gain bearer capabilities support.
In case of a SDCCH a TCH/F will be attempted to be assigned.
This is an open issue for both modes and there should be a
preference for full or half-rate channels somewhere.
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Implement sending MDCX on the newly allocated channel and send
the data to the same destination as the currently connected one.
This way the receiver can implement RTP RFC Appendix A.1 and
deal with the new source.
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For the LCR rtp-bridge audio should directly flow to the
remote system. In contrast to the original patch audio
will now flow directly from the BTS to the remote system.
This assumes that BTS and the remote system are in the
same network segment and can directly communicate.
There are various limitations in the first iteration of
the implementation:
We could (and in the future) should delay the assignment
but currently we are forced to pick the channel and move
it to the audio state. In case we are located on a SDCCH
we always need to change but if we are on a TCH we could
send the ipa.CRCX and change the audio state a lot later.
The net effect is that the audio codec selection needs to
be done in the NITB code and not in the system connected
to it.
This only works with ip based systems. For E1 systems one
could still use the RTP socket or even try to move this
out of the process.
There is no code for handover handling and it relies on
the remote system dealing with the SSRC change of the
system.
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This adds the protocol definition for the RTP bridge extension
of Andreas Eversberg and bumps the protocol version.
I added the missing mncc mappings from value to string.
[ 5cf8fb10ea3addcae74d37f4dbf1c1be664df53e protocol extension
5dac90de38990b188f499c602bf18a4f232070e8 payload extension]
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Before:
<command id='osmux dummy (on|off)'>
<params>
<param name='osmux' doc='RTP multiplexing' />
<param name='dummy' doc='Enable dummy padding' />
<param name='on' doc='Disable dummy padding' />
<param name='off' doc='(null)' />
</params>
After:
<command id='osmux dummy (on|off)'>
<params>
<param name='osmux' doc='RTP multiplexing' />
<param name='dummy' doc='Dummy padding' />
<param name='on' doc='Enable dummy padding' />
<param name='off' doc='Disable dummy padding' />
</params>
</command>
Note the 'null' string in 'off'. Reported by Holger.
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Mike's patch included clean-ups I want to apply separately and
change them a bit. If we return from an else we don't need to
put the else.
* Try the E1 trunk first
* Then try a local virtual trunk
* Fail if none of the above returned
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Remove the host portion of the endpoint Id. This requires less
configuration and we are probably fine to trust that MGCP only
received messages designated for it.
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In the 1900 MHz PCS band, we always generated neighbor cell lists
consisting of only a single neighbor cell, rather than al the
configured BTSs.
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When using multiple interfaces on a system one can now configure
which will be served for the BTS ports and which will be served
for the network. The direct usage of source_addr is now only to
initialize the MGCP receiving port itself.
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Make it possible to bind the call-agent to a specific IP address
and the network and bts end to different ip addresses. Begin by
clarifying which source ip address we want to have.
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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.
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Iridium is a satellite network which operates a GPRS-like that allows you to
get speeds up to 128kbit/s. However, it takes from 5 to 6 secs to get the
bandwidth allocated, so the conversation is garbled during the time.
This patch uses the new dummy padding support in libosmo-netif that is
controlled through the osmux osmux_xfrm_input_open_circuit().
This includes a new VTY option for osmux.
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The spec unfortuantely uses both terms and has no consistent spelling,
but in our logging output we can at least try to be consistent.
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I guess none of our users knows what a mi_type=0x02 is, but most would
know what an IMSI or a TMSI is. So let's use the newly introduced
gsm48_mi_type_name() function to fix this.
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For some odd reasons the XID is not a separate SAPI but has been kludged into
the GMM SAPI. This means we ahve to be careful not to dispatch XID frames into
GMM. We do this by introducing an explicit check for UI frames before the
dispatch to GMM.
The previous code already was doing "the right thing" but printed occasional
messages like "gprs_gmm.c:2082 Unknown GSM 04.08 discriminator 0x01: 01 00 0e
00 32 11 03 16 01 90 63 28 0b". Those should be gone after this patch.
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The same radio link timeout value is used for BTS and MS side.
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Traffic cannot sent to BTS, if there is (currently) no logical channel
associated with the transaction.
This happens, if TCH traffic is received from upper layer, but there is
no lchan available before completing immediate assignment, handover or
assignment process.
[hfreyther: The code has not been moved to tch_frame_down
but the issue looks similiar]
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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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Move forward while preserving the legacy handling. Beging to
extract SDP rtpmap information and select codecs atfer this.
It is a foundation we can now build further and better check
ons.
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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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First collect everything we know and the mapping. E.g. a genuis
could remap "3" to "AMR" so we only know the codecs once we are
at the end of the SDP file. Once we have collected everything we
can select the audio codecs. The current code is compatible in
that two codecs will be selected regardless of if they make any
sense or not.
mgcp_set_audio_info could re-use some of our codec information
but then the caller in the MGCP protocol needs to be updated as
well as we use the "I: GSM" information to derive the codec from
there.
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The SDP file handling will get more complicated in terms of
codec selection so let's remove it from the protocol handling
before we start blowing it up in size.
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This struct was removed when we switched to strtok_r for
parsing the data. Remove the left-over.
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The parsing code assumed that there will be a single payload
type and this assumption is clearly wrong. Forward all of the
payload types. The code is still only extracting the first
type from the list. The variable name has been renamed to
reflect this.
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Using the talloc leak report we see that there are some msgb's
that are allocated for SMS but we don't have transactions or
SMS around. We need to improve the name of the messages to
uniquely dscribe where they are from but the obvious leak does
occur in this routine.
The no available transaction id is most likely the case where
we leak memory. This should not occur and shows another issue
with the smsqueue/smpp handling. It doesn't explain the subscr
reference count issue either.
Extract of the leak report:
GSM 04.11 contains 1160 bytes in 1 blocks (ref 0) 0x2517dc0
GSM 04.11 contains 1160 bytes in 1 blocks (ref 0) 0x24b56e0
GSM 04.11 contains 1160 bytes in 1 blocks (ref 0) 0x23e7930
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For debugging it is nice to know how many requests are
pending. Simply count it and print it besides the paging
part.
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At the rhizomatica sites some unauthorized subscribers have a
last used from long time ago but a bigger reference count. This
means that we have an unbalanced subscr_get/subscr_put that we
need to investigate. These two fixes are genuine fixes but are
unlikely to fix the issue I am not understanding yet.
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In case the SMPP payload didn't include the right fields we
would leak the subscr reference count.
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In case the subscriber is currently busy we would omit the
subscr_put. This seems to be very hard to hit as the subscr
need to be active and at the same time be selected for the
purge operation.
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Various paging clean-ups. I couldn't re-produce the assert that
was hit by Rhizomatica in the subscriber code. Let's clean things
up a bit to avoid some of the potential issues this code had.
Always stop paging regardless of where the subscriber is at the
time of the cancellation.
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So in case somebody is starting paging from within a paging
expired callback we would dispatch the paging request right
away with the same failure.
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