Previously payload_type was always hardcoded to 98 for generated rtp
packets from incoming osmux frame.
Change-Id: I5cbeb494a8932953d9fd2dc24dacf8cd97fd84e4
Until this patch, we didn't notify in any way to the RTP reader when an
Osmux frame was lost. Instead, we updated the seq×tamp as if there
was no lost, and as a result the RTP reader would only see a steady
increase of delay every time an osmux frame was lost.
As the batch_factor for the lost packet is unknown, we cannot assume any
number of amr payloads lost, and thus we cannot simply increment seq and
timestamp for a specific amount. Instead, the only viable solution seems
to set the M marker bit in the first rtp packet generated after a
non-consecutive osmux frame is received.
The implementation may act differently with the first generated RTP
packet based on the first osmux seq number used for the stream. In case
0 it's used as first osmux seq number, M will be set depending on
request from original RTP packet having the M bit set. If it's not 0,
the first RTP packer will unconditionally have the M bit. That's not an
issue because it's anyway expect for receiver to sync on the first
packet.
Related: OS#3185
Change-Id: I2efed6d726a1b8e77e686c7a5fe1940d3f4901a7
With old implementation, in conditions with jitter we could end up
scheduling RTP generated packets from two consecutive osmux frames in an
interleaved way (from seq field point of view).
This new implementation should make it easier for any RTP
reader/playback to have better results in those conditions.
Old APIs osmux_xfm_output and osmux_tx_sched are marked as deprecated in
favour of the new one, which has a better control of generated RTP
packets. However, they are still usable despite the implementation changes
done to support the new API.
Related: OS#3180
Change-Id: I4e05ff141eb4041128ae77812bbcfe84ed4c02de
According to RFC4867 (RTP payload format for AMR):
"The RTP header marker bit (M) SHALL be set to 1 if the first frameblock
carried in the packet contains a speech frame which is the first in a
talkspurt. For all other packets the marker bit SHALL be set to zero (M=0)."
This information bit provides a way for the receiver to better
synchronize the delay with ther sender.
This is specially useful if AMR DTX features are supported and
enabled on the sender.
Change-Id: I0315658159429603f1d80a168718b026015060e9
This new function allows you to create a circuit on an existing input handle.
We don't create the circuit anymore from the first packet seen, instead the
client application is in full control of opening and closing the circuit.
This change includes a new feature to pad a circuit until we see the first
packet that contains voice data. This is useful to preallocate bandwidth on
satellite links such as Iridium/OpenPort.
Add this new function to explicitly remove an existing circuit. Thus, the
client application (openbsc) is in full control to release circuits.
Before this patch, the circuit object was added when the first RTP messages was
seen, and it was removed when transforming the list of pending RTP messages to
the Osmux message (once the timer expired).
Moreover, check circuit->nmsgs to account bytes that are consumed by the osmux
header, given that !circuit doesn't mean "this is the first packet" anymore.
Use the new macros to deal with little/big endian. Im a bit
worried to make this change due the little test coverage in
this module but in case of a typo the elements would not be
defined.
This patch adds a new field to the struct osmux_in_handle that allows
you to specify the osmux frame size. If not specified, the default
size assumes your nic uses a mtu of 1500 bytes.
Remove these functions:
- osmux_xfrm_input_get_ccid
- osmux_xfrm_input_register_ccid
The ccid will be managed by the BSC and it will be stored in the
mgcp_endpoint structure.
Also adjust all tests and examples using the API.
This patch cleans up the transmission path for osmux, this involves
the functions that extract the messages from the batch and the one
that reconstruct the timing.
They now take a list that contains the reconstructed RTP messages:
osmux_xfrm_output(osmuxh, &h_output, &list);
osmux_tx_sched(&list, &tv, tx_cb, NULL);
This patch adds the counter field to the osmux header, so we can
reduce the size of the batch even further, eg.
osmuxhdr (ctr=3)
speech
speech
speech
osmuxhdr (ctr=2)
speech
speech
...
The new header is the following:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FT | CTR |F|Q| SeqNR | Circuit ID |AMR-FT |AMR-CMR|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The counter field is 3 bits long, thus, we can batch up to 8
RTP speech frames into one single batch per circuit ID.
I have also removed the RTP marker, since it can be reconstructed
from the AMR information.
Moreover, the entire workflow has been also reworked. Whenever a
packet arrives, we introduce it into the batch list. This batch
list contains a list of RTP messages ordered by RTP SSRC. Then,
once the batch timer expires or the it gets full, we build the
batch from the list of RTP messages.
Note that this allows us to put several speech frame into one
single osmux header without actually worrying about the amount
of messages that we'll receive.
The functions that reconstruct the RTP messages has been also
adjusted. Now, it returns a list of RTP messages per RTP SSRC
that has been extracted from the batch.
This function schedules the transmission of a RTP message that was
obtained from one osmux batch. It takes the time (in microseconds)
after which the message should be transmitted.