| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
Back in the day we worked around a field proven jitter buffer by
transcoding AMR to AMR with the MTN4200 and by this fixing RTP
timing as this device had:
a.) A working and stable time (generating RTP every N ms)
b.) A working jitter buffer
The code has probably rotted, it never retried MGCP commands sent
to the transcoder and it is not deployed either. Simplify the code
by removing it. If we ever need it back we can use the new transcoding
API to keep the state there. I think this code will not be missed
but the PerformanceTechnology Hard- and Firmware was rather nice.
|
|
Now they are included in a struct we can just call them setup, process
and get_net_downlink_format to manage the lifetime.
|
|
We will have the NOOP implementation (e.g. used by the NAT), the
SW implementation (using software codecs), a HW assisted one that
will use a DSP to do transcoding and in theory the RTP based one
(but I will remove that code).
|
|
The log message does not help and says where the data is
being sent to. This is because we have both a RTP and RTCP
port. Remember if we failed with RTCP or RTP and improve
the log message.
I was searching a case where the port was bound to a local
address (e.g. 127.0.0.1) and tried to send the data to a
public one (e.g. 8.8.8.8).
|
|
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.
|
|
We depend on libosmo-netif unconditionally. Let's use this
definition of rtp and have one portability issue less.
|
|
Honor the IP_TOS settings for Osmux as well. Re-use the RTP
setting as it makes sense to classify the audio packets the
same way.
Fixes: OW#1369
|
|
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.
|
|
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.
|
|
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.
|
|
|
|
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.
|
|
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.
|
|
|
|
|
|
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 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
|
|
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
|
|
|
|
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.
|
|
So far, the jitter computation has been based on output timestamps.
This patch uses the input timestamps instead and resets jitter
computation on SSRC changes.
Sponsored-by: On-Waves ehf
|
|
Currently, when the SSRC changes within a stream and SSRC fixing is
enabled, the RTP timestamp between the last packet that has been
received with the old SSRC and the first packet of the new SSRC
is always incremented by one packet duration.
This can lead to audio muting (at least with the nanoBTS) when the
wallclock interval between these packets is too large (> 1s).
This patch changes the implementation to base the RTP timestamp offset
on the wallclock interval that has passed between these two packets.
Ticket: OW#466
Sponsored-by: On-Waves ehf
|
|
Currently micro-secs and RTP rate get mixed when the transit value is
computed in mgcp_patch_and_count().
This patch changes get_current_ts() to accept the desired rate as
argument and to use it for the time conversion instead of always
converting to microseconds. If microseconds are needed,
get_current_ts(1000) can be used.
The arrival_time is now measured in 1/rtp_end->rate seconds so that
it can be directly compared to RTP timestamps as required by RFC3550
(section 6.4.1, see definition of 'interarrival jitter').
Sponsored-by: On-Waves ehf
|
|
This patch (hopefully) fixes the new defects reported by coverity.
Addresses:
** CID 1156986: Negative array index read (NEGATIVE_RETURNS)
/tests/gsm0408/gsm0408_test.c: 419 in test_si_range_helpers()
/tests/gsm0408/gsm0408_test.c: 423 in test_si_range_helpers()
/tests/gsm0408/gsm0408_test.c: 427 in test_si_range_helpers()
** CID 1156987: Unchecked return value from library
(CHECKED_RETURN)
/src/libmgcp/mgcp_protocol.c: 1150 in mgcp_keepalive_timer_cb()
** CID 1156988: Unchecked return value from library
(CHECKED_RETURN)
/src/libmgcp/mgcp_protocol.c: 983 in handle_modify_con()
Sponsored-by: On-Waves ehf
|
|
Currently a dummy packet is only sent to the RTP port. This is not
enough if RTCP must also cross the SNAT.
This patch sends an additional dummy packet to the RTCP net
destination if omit_rtcp is not set.
Sponsored-by: On-Waves ehf
|
|
Rename the timestamp variable to make in clear, that the input
timestamp is meant. Add a helper variable to illustrate the offset
computation.
Sponsored-by: On-Waves ehf
|
|
This patch make it possible to have a valid endpoint that drops all
outgoing RTP packets. The number of dropped packets is shown by the
VTY 'show mgcp' command. By default, this feature is disabled. To
enable packet dropping, the corresponding output_enabled field must
be set to 0.
Ticket: OW#1044
Sponsored-by: On-Waves ehf
|
|
Currently there are two symmetric code paths which are selected by
the packet destination (NET or BTS).
This patch introduces 3 variables that take the different values
and unifies both code paths into one.
Sponsored-by: On-Waves ehf
|
|
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
|
|
This patch parses the 'ptime' and 'maxptime' SDP attributes, and the
SDP rate information and sets up packet_duration_ms accordingly. If
the packet duration is unknown or allows for different values (e.g.
because 'ptime' uses a range or 'maxptime' allows for more than one
frame) the duration is set to 0.
Sponsored-by: On-Waves ehf
|
|
Currently the timestamp offset calculation is done in two different
places.
This patch moves and unifies both code parts into a separate function.
Sponsored-by: On-Waves ehf
|
|
Since the packet duration is given in ms with the 'ptime' RTP media
attribute and also with the 'p' MGCP local connection option, the
computation is changed to use this value (if present). The
computation assumes, that there are N complete frames in a packet and
takes into account, that the ptime value possibly had been rounded
towards the next ms value (which is never the case with a frame length
of exact 20ms).
Sponsored-by: On-Waves ehf
|
|
This forces the output timing to fulfill
dTS = dSegNo * fixedPacketDuration
where dSegNo = seqNo - lastSeqNo.
If timestamp patching is enabled, the output timestamp will be set
to lastTimestamp + dTS. This kind of relative updating is used to
handle seqNo- and timestamp-wraparounds properly.
The updating of timestamp and SSRC has been separated and the patch
field of mgcp_rtp_state has been renamed to patch_ssrc to reflect
it's semantics more closely. The offset fields are now used always
and will change the corresponding header field if they are != 0.
Ticket: OW#1065
Sponsored-by: On-Waves ehf
|
|
Currently the output SSRC is always forced to be the same if SSRC
patching is enabled.
This patch modifies this to optionally restrict the number of SSRC
changes that will be corrected.
Note that the configuration only allows for the 'once' mode and 'off'.
Sponsored-by: On-Waves ehf
|
|
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
|
|
This patch adds a packet_duration field to mgcp_rtp_state which
contains the RTP packet's duration in RTP timestamp units or 0, when
the duration is unknown or not fixed.
Sponsored-by: On-Waves ehf
|
|
Currently seq_offset and timestamp_offset are updated on each SSRC
change even when SSRC patching is not allowed.
This patch fixes this by changing mgcp_patch_and_count() to only
update these fields when SSRC patching is allowed.
Sponsored-by: On-Waves ehf
|
|
Show old and new SSRC. Move logging command upward to show the values
immediately after the change has been detected and before any fixing
attempt is made.
Sponsored-by: On-Waves ehf
|
|
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
|
|
The tsdelta computation and error detection didn't handle the
intialisation phase properly.
This patches fixes this by skipping the output timing validation
when the SSRCs don't match.
Sponsored-by: On-Waves ehf
|
|
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
|
|
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
|
|
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
|
|
This patch is a cleanup.
|
|
Functions and constants that belong to the libmgcp scope are prefixed
with mgcp_ and MGCP_. This patch is a cleanup.
|
|
This patch replaces the field 'is_transcoded' in the mgcp_endpoint
structure by the enum mgcp_type, that can be further extended with
new types.
|
|
GCC 3.x on PowerPC correctly highlights that the code is fishy.
Re-reading the RFC 3550 shows that we should subtract it and then
we are in the 16bit range. The probation and re-sync code is still
missing.
GCC:
mgcp/mgcp_network.c:200: warning: comparison is always true due to limited range of data type
|
|
The log statement is split into two because inet_ntoa works on an
internal buffer and would print the last address twice.
|
|
|
|
Use a usec timestamp for the local time. The seconds to usec will
swap over to the lower bits but this appears to be correct. The
CLOCK_MONOTONIC is used to fulfill the RFC 3550 requirement even
if it is a bit slower than the gettimeofday.
Make sure to initialize transit in a way that the first transit
time will be 0. Otherwise the jitter will contain the difference
of the localtime and the remote time.
|
