Commit graph

944 commits

Author SHA1 Message Date
George Kiagiadakis
51edc07127 rtprtxsend: use a GSequence to implement the buffer queue
This has the advantage that searching the queue to find the
buffer with the requested seqnum is done with binary search.
2014-01-03 20:48:28 +01:00
George Kiagiadakis
487fa8c989 rtprtxsend: retransmit packets in the same order as the rtx requests 2014-01-03 20:48:28 +01:00
George Kiagiadakis
7d530ab59f rtprtxsend: Handle the max_size_time property
This property allows you to specify the amount of buffers
to keep in the retransmission queue expressed as time (ms)
instead of buffer count (which is the max_size_buffers property).
2014-01-03 20:48:28 +01:00
George Kiagiadakis
920a55532c rtprtxsend: keep important buffer information in a private structure
This is to avoid mapping a buffer every time we need to read a seqnum
or a timestamp.
2014-01-03 20:48:28 +01:00
Julien Isorce
5a1aa75961 rtpmanager: add new rtprtxsend / rtprtxreceive elements
The purpose of the sender RTX object is to keep a history
of RTP packets up to a configurable limit (in time). It will
listen for custom retransmission events from downstream. When
it receives a request for retransmission, it will look up the
requested seqnum in its list of stored packets. If the packet
is available, it will create a RTX packet according to RFC 4588
and send this as an auxiliary stream.

The receiver will listen to the custom retransmission events
from the downstream jitterbuffer and will remember the SSRC1
of the stream and seqnum that was requested. When it sees a
packet with one of the stored seqnum, it associates the SSRC2
of the stream with the SSRC1 of the master stream. From then
on it knows that SSRC2 is the retransmission stream of SSRC1.
This algorithm is stated in RFC 4588. For this algorithm to
work, RFC4588 also states that no two pending retransmission
requests can exist for the same seqnum and different SSRCs or
else it would be impossible to associate the retransmission with
the original requester SSRC.
When the RTX receiver has associated the retransmission packets,
it can depayload and forward them to the source pad of the element.

RTX is SSRC-multiplexed

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
2014-01-03 20:47:59 +01:00
Wim Taymans
bb2d37b11d rtpbin: add some docs about AUX elements 2013-12-31 15:08:49 +01:00
Wim Taymans
d08e05b4ef rtpbin: add support for AUX sender and receiver
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.

The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
2013-12-31 15:08:48 +01:00
Wim Taymans
ae22c95881 rtpbin: make request_element method internally
We can use the same method to create encoder and decoder elements, they
are just internal elements that we create.
2013-12-31 15:08:48 +01:00
Wim Taymans
ee7f41ba2e rtpsession: internal-ssrc is no longer deprecated 2013-12-30 17:00:45 +01:00
Wim Taymans
e721d26c68 rtpbin: add Since tags 2013-12-30 16:59:20 +01:00
Wim Taymans
5a2bc1405e rtpbin: add signal for new jitterbuffer
Emit a signal when a new jitterbuffer is created so that the app can
have a chance to configure it.
2013-12-30 16:52:28 +01:00
Wim Taymans
3f3b2d0886 rtpbin: handle multiple encoder instances
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
2013-12-30 16:28:57 +01:00
Wim Taymans
05c8edc174 rtpbin: fix memory leaks 2013-12-30 15:17:05 +01:00
Wim Taymans
9345c2280a rtpbin: expect the pads on the encoders
Don't use request pads for the encoder elements, the signal handler
should request the pads and make sure they are available with the right
name.
2013-12-30 15:17:05 +01:00
Wim Taymans
cbc80d10dd rtpbin: request-rtp-encoder are no action signals
The request-rtp-encoder signals are not action signals so mark them
correctly and use an accumulator to collect the result value.
2013-12-30 15:17:05 +01:00
George Kiagiadakis
5ddf6a5e32 gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
2013-12-30 14:03:05 +01:00
George Kiagiadakis
17517ca491 rtpsession: allow setting internal-ssrc again 2013-12-30 14:03:05 +01:00
Aleix Conchillo Flaqué
47c65fc269 rtpbin: allow dynamic RTP/RTCP encoders/decoders
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
  added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
  and request-rtcp-decoder). The user will be able to provide encoders
  or decoders dynamically. The encoders must follow the srtpenc API and
  the decoders the srtpdec API. Having separate signals for RTP and RTCP
  allows the user to use different encoders/decoders or provide the same
  one (e.g. that would be the case for srtpenc).

  Also, rtpbin now allows application/x-srtp in its pads.

  https://bugzilla.gnome.org/show_bug.cgi?id=719938
2013-12-30 11:24:00 +01:00
Wim Taymans
f48bbabafc rtpjitterbuffer: dynamically recalculate RTX parameters
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.

Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
2013-12-30 11:18:51 +01:00
Wim Taymans
416bd9a2c3 rtpjitterbuffer: calculate average jitter 2013-12-30 11:18:51 +01:00
Wim Taymans
7181a21ca9 rtpsession: use RTT from the Retransmission event
Place the estimated RTT in the Retransmission event and let the session
manager use that instead of the hardcoded value.
2013-12-30 11:18:50 +01:00
Wim Taymans
e996f73d0c jitterbuffer: take more accurate running-time for NACK
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
2013-12-30 11:18:50 +01:00
Olivier Crête
ada6ea668b rtpsession: Add error message if the app tries to set the internal-ssrc 2013-12-13 17:36:36 -05:00
Olivier Crête
d715010d78 rtpsession: Only count nacks when a nack packet is received
Not when any RTCP feedback packet is.
2013-12-13 16:08:35 -05:00
Olivier Crête
7af9fdbca6 rtpsession: Process PSFB FIR requests which lack the media ssrc
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.

Fixes a regression introduced by commit 57c27ec3
2013-12-13 16:01:07 -05:00
George Kiagiadakis
6a2de911fa rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.

This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
2013-12-12 16:44:27 +01:00
George Kiagiadakis
c78a115154 rtpsession: keep extra stats for scheduling BYE
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
2013-12-12 10:38:43 +01:00
George Kiagiadakis
282028e753 rtpsession: when we schedule BYE, only deal with BYE sources
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
2013-12-12 10:34:38 +01:00
George Kiagiadakis
6a421c3d81 rtpsession: reset state after scheduling BYE
After we do RTCP, we are not scheduling bye anymore.
2013-12-12 10:32:48 +01:00
George Kiagiadakis
0a0ff100ef rtpsession: also count NACKS when no signal was pending 2013-12-12 10:31:38 +01:00
George Kiagiadakis
bec9c04ea0 session: ignore RTCP packets for the BYE sources
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
2013-12-12 10:09:25 +01:00
Julien Isorce
33b398e345 rtpsession: determine if the session is doing point-to-point
In this case T_dither_max is set to 0 according to RFC 4585
2013-12-10 16:57:56 +01:00
Wim Taymans
eee515cb2c rtpjitterbuffer: serialize events in the buffer
Serialize events into the jitterbuffer by inserting them with a -1
seqnum.
Update unit test to expect events from the streaming thread.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
2013-12-10 11:57:37 +01:00
Wim Taymans
36e78bc5ca rtpjitterbuffer: detect -1 seqnum
Keep the seqnum as a full guint so that we can check for -1 entries and
deal with them correctly.
Immediately try to push -1 seqnum.
2013-12-10 11:04:06 +01:00
Wim Taymans
4a2e0f4ff4 rtpjitterbuffer: reorganize jitterbuffer items
Keep the oldest item at the head and the newest items on the tail. This
makes it easier to deal with -1 seqnums.
2013-12-10 11:01:03 +01:00
Wim Taymans
ea2a222cef jitterbuffer: correctly check for invalid values
Check for -1 on the guint from the buffer item instead of on the guint16
or guint32.
Also insert -1 seqnum at the head of the jitterbuffer.
2013-12-09 23:34:10 +01:00
Wim Taymans
e8edecc56e rtpsession: don't unref buffer twice
Cleaning the packet info will already unref the buffer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078
2013-11-28 16:51:13 +01:00
Wim Taymans
710d1f3a2a rtpjitterbuffer: improve clear-pt-map handling
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
2013-11-25 15:52:22 +01:00
Tim-Philipp Müller
901ec63462 rtpjitterbuffer: fix wake-up when new buffers come in after running empty
Spotted by 'gratias' on IRC. Probably introduced in recent refactoring.

https://bugzilla.gnome.org/show_bug.cgi?id=715039
2013-11-25 00:37:50 +00:00
Wim Taymans
4c9474905b rtpjitterbuffer: pass downstream flowreturn to upstream
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722
2013-11-22 12:27:31 +01:00
Wim Taymans
0c6f4efe4a rtpjitterbuffer: avoid mapping the buffer
Reuse the parsed structure to get the timestamps.
2013-11-19 10:12:00 +01:00
Tim-Philipp Müller
d506409af5 docs: get rid of 'Since: 0.10.x' markers
And some gtk-doc markup fixes.
2013-11-18 14:47:35 +00:00
Tim-Philipp Müller
548e756e0a rtpmanager: fix Since markers
Should be next stable release series version
2013-11-16 12:15:14 +00:00
George Kiagiadakis
387e3b918a rtpjitterbuffer: Fix stats property field names and documentation 2013-11-15 16:23:34 +02:00
Torrie Fischer
acf74435e3 gstrtpsession: Implement a number of feedback packet statistics
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693
2013-11-15 15:21:19 +01:00
Wim Taymans
b450d31503 rtpjitterbuffer: rename property to 'stats'
This makes the unit test work.
We can later also add more stats, not specific to retransmission.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411
2013-11-14 09:24:26 +01:00
Torrie Fischer
22ceb80ba9 rtpjitterbuffer: implement rtx statistics 2013-11-14 09:24:26 +01:00
Wim Taymans
2e5b462ae3 jitterbuffer: advance expected seqnum after dropping
After dropping a buffer, move our expected seqnum

Conflicts:
	gst/rtpmanager/gstrtpjitterbuffer.c
2013-11-13 12:02:57 +01:00
Wim Taymans
e4bc81d7d2 rtpsession: remove collision reconfigure event
Remove bogus reconfigure event on collision, we don't want to send the event on
the receiving RTP pad and the collision event is now handling this
case.

See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:27:18 +01:00
Julien Isorce
b32fc6f416 gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad
See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:25:52 +01:00
George Kiagiadakis
b81b2efa3e rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost
The problem here was that the jitterbuffer lock was unlocked to push
the event, but that caused another thread to remove the timer currently
being processed, probably because the amount of rtx events
(and therefore timers) was getting too high. The solution is to
unlock and push the event only after timer processing has finished.

fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131
2013-11-11 11:51:45 +01:00
Wim Taymans
c8db05d610 rtpsource: update receiver stats for sender
An internal sender in a session is also a receiver of its own packets so update
the receiver stats. Other senders in the session will use this info to generate
correct RB blocks in their SR reports.
2013-11-07 16:24:30 +01:00
Wim Taymans
268a75e705 rtpsource: refactor receiver stats update 2013-11-07 16:24:30 +01:00
Wim Taymans
e96f8f519c rtspsrc: proxy new buffer mode 2013-10-31 10:38:35 +01:00
Wim Taymans
43645d5981 jitterbuffer: add new timestamp mode
Add a new timestamp mode that assumes the local and remote clock are
synchronized. It takes the first timestamp as a base time and then uses the RTP
timestamps for the output PTS.
2013-10-31 10:15:25 +01:00
Wim Taymans
d4892859d4 jitterbuffer: fix race in flush-start/flush-stop
When flush-stop arrives before we process the result of the _push() in the
loop function, we might pause even though we are not flushing anymore. Fix this
race by waiting for the srcpad loop function to completely pause after doing the
flush-start.
2013-10-04 12:35:18 +02:00
Wim Taymans
00056965e8 rtpjitterbuffer: fix race when updating the next_seqnum
If we were not waiting for the missing seqnum when we insert the lost packet
event in the jitterbuffer, we end up not updating the next_seqnum and wait
forever for the lost packets to arrive. Instead, keep track of the amount of
packets contained by the jitterbuffer item and update the next expected
seqnum only after pushing the buffer/event. This makes sure we correctly handle
GAPS in the sequence numbers.
2013-09-30 12:31:00 +02:00
Wim Taymans
fde438791e rtpjitterbuffer: small debug improvement 2013-09-30 12:30:23 +02:00
Wim Taymans
6e7d547be4 rtpjitterbuffer: reset skew does not reset clock-rate
Don't reset the clock-rate when we reset the skew correction algorithm.
Reset the skew correction algorithm when we change the clock-rate.
2013-09-30 11:53:08 +02:00
Wim Taymans
03d520eb69 rtpjitterbuffer: pause timer when PAUSED
Also pause the timer when we go to the PAUSED state. It is possible that we
don't have a clock or base-time in PAUSED to perform the timeouts.
2013-09-30 11:16:32 +02:00
Wim Taymans
4a31aec513 rtpjitterbuffer: improve debug 2013-09-30 11:15:25 +02:00
Wim Taymans
d4b4b4e924 rtpjitterbuffer: don't calculate skew without rtptime
Skip trying to calculate the skew when we don't have an rtptime.
It causes problems when lost packet events are placed in the jitterbuffer.
2013-09-26 16:21:33 +02:00
Wim Taymans
2efd58fc84 rtpbin: avoid some pad link checks
Link pads without checks, we know it will work.
2013-09-25 17:38:31 +02:00
Wim Taymans
97f4674655 rtpjitterbuffer: calculate some stats 2013-09-25 10:50:05 +02:00
Wim Taymans
b1d29483bb rtpjitterbuffer: move send_lost_event function
Move the send_lost_event function to the do_lost_event handling, there is no
need to have a separate function.
2013-09-25 10:50:05 +02:00
Wim Taymans
adf5d96044 rtpmanager: update docs 2013-09-23 16:34:15 +02:00
Wim Taymans
e5019de80d docs: update docs with 1.0 element names 2013-09-23 15:36:47 +02:00
Wim Taymans
8ce674da87 rtpjitterbuffer: always store lost event in jitterbuffer
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
2013-09-23 14:45:27 +02:00
Wim Taymans
9f3345fcc2 rtpjitterbuffer: schedule lost event differently
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
2013-09-23 14:45:27 +02:00
Wim Taymans
ae389aeb0c rtpjitterbuffer: remove list debug 2013-09-23 14:45:26 +02:00
Wim Taymans
28641e3145 rtpjitterbuffer: add type to the item
So that the upper layer can know what data is contained in the item.
2013-09-23 14:45:26 +02:00
Wim Taymans
479c7642fd rtpjitterbuffer: fix flush
Pass function to flush to properly free the queue items.
2013-09-23 14:45:25 +02:00
Wim Taymans
0cc887eb98 rtpjitterbuffer: append seqnum -1 packets 2013-09-23 14:45:25 +02:00
Wim Taymans
39a2ba669d rtpjitterbuffer: use structure to hold packet information
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
2013-09-23 14:45:25 +02:00
Wim Taymans
1760817005 rtpjitterbuffer: update expected timer when possible
When we receive a packet and we have some missing packets, we can update their
estimated arrival times based on the timestamp difference.
2013-09-23 14:45:25 +02:00
Wim Taymans
fdc1ed1680 rtpjitterbuffer: fix order of timeout events
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
timeout.
2013-09-23 14:45:25 +02:00
Wim Taymans
0b1a7edfea rtpjitterbuffer: send lost event before signaling next buffer
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
2013-09-23 14:45:25 +02:00
Wim Taymans
5051f51f0a jitterbuffer: configure clock-rate on jitterbuffer
Add a get and setter to configure the clock-rate in the jitterbuffer instead of
passing it as an argument to the insert method.
2013-09-23 14:45:24 +02:00
Wim Taymans
3c421e7e48 rtpjitterbuffer: add option to reset retransmission timers 2013-09-23 14:45:24 +02:00
Wim Taymans
6f4deab298 rtpjitterbuffer: stop the timer thread
The timeout code could release the lock so we need to check if we are allowed to
wait for the clock some more.
2013-09-23 14:45:24 +02:00
Wim Taymans
cba4e6a707 rtpjitterbuffer: unlock only once 2013-09-23 14:45:24 +02:00
Wim Taymans
5dc207948c rtpjitterbuffer: improve flush and shutdown
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
2013-09-23 14:45:23 +02:00
Wim Taymans
a512cc2d3c rtpjitterbuffer: set correct expected time
When we already have a timer for a packet, skip it but don't forget to adjust
the dts to the expected dts of the next packet.
2013-09-23 14:45:23 +02:00
Wim Taymans
517ea0f4d9 jitterbuffer: improve debug 2013-09-23 14:45:23 +02:00
Wim Taymans
fd6c57cf10 rtpjitterbuffer: keep delay as a separate variable in timer
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
2013-09-19 14:32:48 +02:00
Wim Taymans
d34184dd03 rtpjitterbuffer: fix writability of properties 2013-09-19 14:32:48 +02:00
Wim Taymans
6bb2626498 rtpjitterbuffer: reevaluate the current timer after timeout
When we trigger the timeout logic of a timer, reevaluate it because it is
possible that it still has the lowest timeout.
2013-09-18 16:33:40 +02:00
Wim Taymans
8d021b6ede rtpjitterbuffer: don't update time when unscheduled
Don't try to estimate the current time when we got unscheduled.
2013-09-18 16:31:26 +02:00
Wim Taymans
65606a25bf rtpjitterbuffer: init packet spacing on first buffer
Already init the packet spacing variables on the first buffer so that we can
calculate the spacing on the second buffer already.
2013-09-18 16:29:37 +02:00
Wim Taymans
f2efdf28f5 rtpjitterbuffer: push the lost event from the timer thread
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
2013-09-18 14:57:09 +02:00
Wim Taymans
5d5fc03e04 rtpjitterbuffer: round gap duration to multiple of duration
Make sure the gap duration in the lost event is a multiple of the packet
duration.
Enable another test.
2013-09-18 14:12:47 +02:00
Wim Taymans
6e4a051d40 rtpjitterbuffer: keep track of duration
Keep track of the estimated duration of missing packets and use it in the lost
event.
Enable another unit test
2013-09-18 12:29:38 +02:00
Wim Taymans
ac3bb3acf6 rtpjitterbuffer: handle large gaps with one lost event
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
2013-09-18 11:59:28 +02:00
Wim Taymans
26402e1c21 rtpjitterbuffer: refactor lost event sending
Also make sure we only increment the expected seqnum and last
output timestamp.
2013-09-18 11:57:06 +02:00
Wim Taymans
f49981a597 jitterbuffer: refactor timeout triggers 2013-09-17 23:29:56 +02:00
Wim Taymans
047021c443 jitterbuffer: simplify the timeout code
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
2013-09-17 23:29:56 +02:00
Wim Taymans
fa1ef3328b jitterbuffer: rearrange timer update code
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
2013-09-17 23:29:56 +02:00
Wim Taymans
232fdd8b56 jitterbuffer: release lock on shutdown 2013-09-17 15:19:42 +02:00
Wim Taymans
4de919a17a jitterbuffer: use separate thread for timeouts
Use a separate thread for scheduling the timeouts instead of using the
downstream streaming thread that might block at any time.
2013-09-16 15:55:55 +02:00
Olivier Crête
b9ceafe5af rtpsession: Demux RTCP buffers from the RTP stream
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761

https://bugzilla.gnome.org/show_bug.cgi?id=687657
2013-09-13 16:25:49 +02:00