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The bridge_p2p_rtp_write() has potential reentrancy problems.
* Accessing the bridged RTP members must be done with the instance1 lock
held. The DTMF and asymmetric codec checks must be split to be done with
the correct RTP instance struct locked. i.e., They must be done when
working on the appropriate side of the point to point bridge.
* Forcing the RTP mark bit was referencing the wrong side of the point to
point bridge. The set mark bit is used everywhere else to set the mark
bit when sending not receiving.
The patches for ASTERISK_26745 and ASTERISK_27158 did not take into
account that not everything carried by RTP uses a codec. The telephony
DTMF events are not exchanged with a codec. As a result when
RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is
enabled, the DTMF digits would always get passed to the core even though
the local native RTP bridge is active, and the DTMF digits would go out
using the wrong SSRC id.
* Add protection for non-format payload types like DTMF when updating the
lastrxformat and lasttxformat. Also protect against non-format payload
types when checking for asymmetric codecs.
ASTERISK-27292
Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186
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Validate RTCP packets before processing them.
* Validate that the received packet is of a minimum length and apply the
RFC3550 RTCP packet validation checks.
* Fixed potentially reading garbage beyond the received RTCP record data.
* Fixed rtp->themssrc only being set once when the remote could change
the SSRC. We would effectively stop handling the RTCP statistic records.
* Fixed rtp->themssrc to not treat a zero value as special by adding
rtp->themssrc_valid to indicate if rtp->themssrc is available.
ASTERISK-27274
Make strict RTP learning more flexible.
Direct media can cause strict RTP to attempt to learn a remote address
again before it has had a chance to learn the remote address the first
time. Because of the rapid relearn requests, strict RTP could latch onto
the first remote address and fail to latch onto the direct media remote
address. As a result, you have one way audio until the call is placed on
and off hold.
The new algorithm learns remote addresses for a set time (1.5 seconds)
before locking the remote address. In addition, we must see a configured
number of remote packets from the same address in a row before switching.
* Fixed strict RTP learning from always accepting the first new address
packet as the new stream.
* Fixed strict RTP to initialize the expected sequence number with the
last received sequence number instead of the last transmitted sequence
number.
* Fixed the predicted next sequence number calculation in
rtp_learning_rtp_seq_update() to handle overflow.
ASTERISK-27252
Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c
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Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998
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Change-Id: Ic6493a7d79683f3e5845dff1cee49445fd5a0adf
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This change moves the logic which learns a new source address
for RTP so it only occurs in the learning state. The learning
state is entered on initial allocation of RTP or if we are
told that the remote address for the media has changed. While
in the learning state if we continue to receive media from
the original source we restart the learning process. It is
only once we receive a sufficient number of RTP packets from
the new source that we will switch to it. Once this is done
the closed state is entered where all packets that do not
originate from the expected source are dropped.
The learning process has also been improved to take into
account the time between received packets so a flood of them
while in the learning state does not cause media to be switched.
Finally RTCP now drops packets which are not for the learned
SSRC if strict RTP is enabled.
ASTERISK-27013
Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c
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Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
ASTERISK-26745 #close
Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
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Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated. Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.
ASTERISK-27158 #close
Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
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Change-Id: Ia578ede1a55b21014581793992a429441903278b
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This change makes it so that if an RTCP packet is being sent
the RTP ICE component is used for sending if RTCP-MUX is in use.
ASTERISK-27133
Change-Id: I6200f611ede709602ee9b89501720c29545ed68b
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When a message is received on the TURN socket, the code processing the
message needs to call into the ICE/STUN session for further processing.
This code path locks the TURN group lock then the ICE/STUN group lock. In
another thread an ICE/STUN timer can fire off to send a keep alive message
over the TURN socket. In this code path, the ICE/STUN group lock is
obtained then the TURN group lock is obtained to send the packet. A
classic deadlock case if the group locks are not the same.
* Made TURN get created using the ICE/STUN session's group lock.
NOTE: I was originally concerned that the ICE/STUN session can get
recreated by ice_reset_session() for an event like RTCP multiplexing
causing a change during SDP negotiation. In this case the TURN group lock
would become different. However, TURN is also recreated as part of the
ICE/STUN recreation in ice_create() when all known ICE candidates are
added to the new ICE session. While the ICE/STUN and TURN sessions are
being recreated there is a period where the group locks could be
different.
ASTERISK-27023 #close
Patches:
res_rtp_asterisk-turn-deadlock-fix.patch (license #6502)
patch uploaded by Michael Walton (modified)
Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9
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When re-inviting to add more streams it is possible for
the role of existing ICE sessions to be changed to the
incorrect value. This results in subsequent refreshes
within the sessions getting a role conflict and the ICE
session breaking down. This change only sets the role to
be the new value if an ICE renegotiation is actually
going to happen, otherwise the existing role is preserved.
As well if we encounter a situation where a unidirectional
ICE negotiation happens and the other side does not send us
candidates we will not store any information for sending
traffic, even though we know where they are reachable. This
change fixes this by using the source of the ICE traffic
itself as the target if no candidates are known and we
receive some ICE traffic.
ASTERISK-27088
Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9
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It looks like there was a copy/paste error in ast_rtp_change_source
where if there was a rtcp srtp instance, instead of updating its
ssrc we were updating the srtp instance ssrc twice.
ASTERISK-27022 #close
Reported-by: Michael Walton
Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095
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In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
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When using rtcp mux if an rtcp payload came in it would still use the srtp
unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp
data was being passed to the rtp unprotect method this would result in an
error.
This patch ensures that the correct unprotect method is chosen by making
sure the passed in rtcp flag is appropriately set when rtcp mux is enabled
and an rtcp payload is received.
ASTERISK-26979 #close
Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241
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When a call gets put on hold RTP is temporarily stopped and Asterisk was
setting the remote RTCP address to NULL. Then when RTCP data was received
from the remote endpoint, Asterisk would be missing this information when
publishing the rtcp_message stasis event. Consequently, message subscribers
(in this case res_hep_rtcp) trying to parse the "from" field output the
following error:
"ast_sockaddr_split_hostport: Port missing in (null)"
This patch makes it so the remote RTCP address is no longer set to NULL when
stopping RTP. There was only one place that appeared to check if the remote
RTCP address was NULL as a way to tell if RTCP was running. This patch added
an additional check on the RTCP schedid for that case to make sure RTCP was
truly not running.
ASTERISK-26860 #close
Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b
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Occasionally a crash happens when processing the RTCP DTLS timeout
handler. The RTCP DTLS timeout timer could be left running if we have not
completed the DTLS handshake before we place the call on hold or we
attempt direct media.
* Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP.
* Made some sanity tweaks to ast_rtp_prop_set() when switching from
standard RTCP mode to RTCP multiplexed mode.
ASTERISK-26692 #close
Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0
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The struct ast_rtp_instance has historically been indirectly protected
from reentrancy issues by the channel lock because early channel drivers
held the lock for really long times. Holding the channel lock for such a
long time has caused many deadlock problems in the past. Along comes
chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock
because sometimes there may not be an associated channel created yet or
the channel pointer isn't available.
In the case of ASTERISK-26835 a pjsip serializer thread was processing a
message's SDP body while another thread was reading a RTP packet from the
socket. Both threads wound up changing the rtp->rtcp->local_addr_str
string and interfering with each other. The classic reentrancy problem
resulted in a crash.
In the case of ASTERISK-26853 a pjsip serializer thread was processing a
message's SDP body while another thread was reading a RTP packet from the
socket. Both threads wound up processing ICE candidates in PJPROJECT and
interfering with each other. The classic reentrancy problem resulted in a
crash.
* rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP
instance struct.
* rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP
instance struct for the API call.
* res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy
problem with rtp->rtcp->local_addr_str in the scheduler thread running
ast_rtcp_write().
* res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in
bridge_p2p_rtp_write() because there are two RTP instance structs
involved.
* res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler
callbacks. We cannot hold the instance lock when trying to stop a
scheduler callback.
* res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the
struct ast_rtp_instance ao2 object lock instead. The lock was used to
synchronize two threads to prevent a race condition between starting and
stopping a timeout timer. The race condition is no longer present between
dtls_perform_handshake() and __rtp_recvfrom() because the instance lock
prevents these functions from overlapping each other with regards to the
timeout timer.
* res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct
ast_rtp_instance ao2 object lock instead. The lock was used to
synchronize two threads using a condition signal to know when TURN
negotiations complete.
* res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN
ioqueue_worker_thread(). We cannot hold the instance lock when trying to
create or shut down the worker thread without a risk of deadlock.
This patch exposed a race condition between a PJSIP serializer thread
setting up an ICE session in ice_create() and another thread reading RTP
packets.
* res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we
have re-locked the RTP instance to prevent the other thread from trying to
process ICE packets on an incomplete ICE session setup.
A similar race condition is between a PJSIP serializer thread resetting up
an ICE session in ice_create() and the timer_worker_thread() processing
the completion of the previous ICE session.
* res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an
uninitialized/null remote_address after calling
update_address_with_ice_candidate().
* res_rtp_asterisk.c: Eliminate the chance of ice_reset_session()
destroying and setting the rtp->ice pointer to NULL while other threads
are using it by adding an ao2 wrapper around the PJPROJECT ice pointer.
Now when we have to unlock the RTP instance object to call a PJPROJECT ICE
function we will hold a ref to the wrapper. Also added some rtp->ice NULL
checks after we relock the RTP instance and have to do something with the
ICE structure.
ASTERISK-26835 #close
ASTERISK-26853 #close
Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4
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Added the stun_blacklist option to rtp.conf. Some multihomed servers have
IP interfaces that cannot reach the STUN server specified by stunaddr.
Blacklist those interface subnets from trying to send a STUN packet to
find the external IP address. Attempting to send the STUN packet
needlessly delays processing incoming and outgoing SIP INVITEs because we
will wait for a response that can never come until we give up on the
response. Multiple subnets may be listed.
ASTERISK-26890 #close
Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342
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We are currently passing in the capacity of the read buffer instead of the
number of bytes that we actually read off the wire.
Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36
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stopped." into 14
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struct ast_rtcp does not define the dtls member if SRTP is not enabled.
ASTERISK-26732
Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e
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This change removes an assumption that when DTLS is stopped
an RTCP session will be present on the RTP session. This is not
always the case.
ASTERISK-26732
Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611
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This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.
A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.
The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.
ASTERISK-26732 #close
Reported by Dan Jenkins
Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
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* Removed all 2.5.5 functional patches.
* Updated usages of pj_release_pool to be "safe".
* Updated configure options to disable webrtc.
* Updated config_site.h to disable webrtc in pjmedia.
* Added Richard Mudgett's recent resolver patches.
Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7
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pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND,
which is a compile-time constant. Instead of hard-coding 16 when we
enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can
potentially collect more interfaces if the compile time options are
changed.
Tangentially related to ASTERISK~24464
Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd
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Before Asterisk 13, signed linear was converted into network byte order by a
smoother before being sent over the network. We restore this behavior by
forcing the creation of a smoother when slinear is in use and setting the
appropriate flags so that the byte order conversion is always done.
ASTERISK-24858 #close
Reported-by: Frankie Chin
Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9
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* channel.c:ast_sendtext(): Fix T.140 SendText memory leak.
* format_compatibility.c: T.140 RED and T.140 were swapped.
* res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak.
* res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic
scheduled red_write().
* res_rtp_asterisk.c: Some other minor misc tweaks.
Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb
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The mechanism used for detecting the maximum log level compiled into the
linked pjproject did not work. The API call simply stores the requested
level into an integer and does no range checking. Asterisk was assuming
that there was range checking and limited the new value to the allowable
range. To get the actual maximum log level compiled into the linked
pjproject we need to get and save off the initial set log level from
pjproject. This is the maximum log level supported.
* Get and save off the initial log level setting before altering it to the
desired level on startup. This has to be done by a macro rather than
calling a core function to avoid incorrectly linking pjproject.
* Split the initial log level warning messages to warn if the linked
pjproject cannot support the requested startup level and if it is too low
to get the pjproject buildopts for "pjproject show buildopts".
* Adjust the CLI "pjproject set log level" to check the saved max log
level and to generate normal output messages instead of a warning message.
ASTERISK-26743 #close
Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4
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ast_rtp_remote_address_set() could pass an uninitialized 'us' parameter to
ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us'
parameter may not get initialized. Thus when the code tries to save the
'us' parameter to the local address we could try to copy a ridiculous
sized memory buffer and segfault.
* Made pass an initialized 'us' parameter to ast_ouraddrfor().
* Optimized out the 'us' struct variable.
ASTERISK-26672 #close
Change-Id: I4acea5dcdf0813da2c7d3e11c2d6067d160d17dc
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We access uninitialized memory when the 'ourip' parameter does not
have an initial guess to our IP address.
ASTERISK-26672
Change-Id: I35507ea1ad7455d2be188f6ccdd4add7bd150e15
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Change-Id: I95b1088d11244a2edae6607c12fbf33b38658a75
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Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.
Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages. Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible. Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.
* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.
* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.
* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.
* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject. Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.
* In log_forwarder(), made always log enabled and mapped pjproject log
messages. DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.
* Removed RAII_VAR() from res_pjproject.c:get_log_level().
ASTERISK-26630 #close
Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
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When retrieving RTCP stats for PJSIP channels, RTT values are unreliable.
RTT calculation is correct, but the data representation isn't. RTT is
represented by a 32-bit fixed-point number with the integer part in the
first 16 bits and the fractional part in the last 16 bits. In order to
get the RTT value, the fractional part is miscalculated, there is an
unnecessary 16 bit shift that causes overflow. Besides this there is
another mistake, when transforming the integer value to the fixed point
fractional part via bitwise operation, that loses precision.
* RTT fractional part is no longer shifted, avoiding overflow.
* RTT fractional part is transformed to its fixed-point value more
precisely.
* Fixed timeval2ntp() and ntp2timeval() second fraction conversions.
* Fixed NTP timestamp report logging. The usec was inexplicably
multiplied by 4096.
ASTERISK-26566 #close
Reported by Hector Royo Concepcion
Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f
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OpenBSD's 'find' doesn't take the -delete argument so you have to pipe
through 'xargs rm -rf'.
'echo -e' doesn't like \t starting a line. It just prints 't' which
causes the libasteriskpj.exports file to be garbage. They were just
cosmetic so they were removed.
librt doesn't exist so the link of libasteriskpj.so fails. It's not
actually needed for linux anyway so -lrt was removed from the link.
res_rtp_asterisk was failing to load because of an undefined
DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if
so DTLSv1_method is used instead.
ASTERISK-26608
Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c
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This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.
ASTERISK-26309 #close
Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d
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Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the
form ice_blacklist = <subnet spec>, e.g. ice_blacklist =
192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay
discovery. This is useful for optimizing the ICE process where a system
has multiple host address ranges and/or physical interfaces and certain
of them are not expected to be used for RTP. Multiple ice_blacklist
configuration lines may be used. If left unconfigured, all discovered
host addresses are used, as per previous behavior.
Documention in rtp.conf.sample.
ASTERISK-26418 #close
Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9
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If a bridge switched to P2P when a DTMF was in progress it
was possible for the DTMF to continue being sent indefinitely.
Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29
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Currently when receiving video over RTP we store only
a calculated samples on the frame. When starting the video
it can take some time for this calculation to actually yield
a value as it requires constant changing timestamps. As well
if a video frame passes over multiple RTP packets this calculation
will fail as the timestamp is the same as the previous RTP
packet and the number of samples calculated will be 0.
This change preserves the timestamp on the frame and allows
it to pass through the core. When sending the video this timestamp
is used instead of a new one being calculated.
ASTERISK-26367 #close
Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
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When an RTCP packet is sent or received, res_rtp_asterisk generates a
Stasis event that contains the RTCP report as well as the local and
remote addresses that the report pertains to.
The addresses are determined using ast_find_ourip(). For the local
address, this will typically result in a lookup of the hostname of the
server, and then a DNS lookup of that hostname. If you do not have the
host in /etc/hosts, then this results in a full DNS lookup, which can
potentially block for some time.
This is especially problematic when performing RTCP reads, since those
are done on the same thread responsible for reading and writing media.
This patch addresses the issue by performing a lookup of the local
address when RTCP is allocated. We then use this cached local address
for the Stasis events when necessary.
ASTERISK-26280 #close
Reported by Mark Michelson
Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556
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With this change, the initial RTP sequence number is randomly chosen not between
0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
counter (ROC) synchronization is not lost for sRTP, when the very first RTP
packets get lost; see http://srtp.sourceforge.net/faq.html#Q6
ASTERISK-26207 #close
Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464
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Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
for DTLS. The source code from main/tcptls.c should have been re-used to ease
security audits. Therefore, this change rolls back the change from July 2015 and
re-uses the code from July 2014. This has the additional benefits to work under
CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.
ASTERISK-25659 #close
Reported by: StefanEng86, urbaniak, pay123
Tested by: sarumjanuch, traud
patches:
res_rtp_asterisk.patch submitted by sarumjanuch
dtls_centos_step_1.patch submitted by traud
dtls_centos_step_2.patch submitted by traud
Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c
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gcc 6 caught a previously unidentified self-comparison in
ice_candidate_cmp. Fixed it and re-ordered the predicates for better
short-circuiting.
ASTERISK-26140 #close
Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7
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ensure that cert bios get freed after creating the fingerprint
ASTERISK-26129 #close
Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451
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Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the
underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for
WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based
cipher-suites.
ASTERISK-26130 #close
Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0
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Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount
for rtcp packets as well as rtp packets and that was causing sender reports
to be generated instead of receiver reports in cases where no rtp was actually
being sent.
Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp,
to rtp_sento which only handles rtp packets.
Discovered by the hep/rtcp-receiver test.
Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5
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Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764
explicitly states:
There MUST be a separate DTLS-SRTP session for each distinct pair of
source and destination ports used by a media session
This means RTP keying material cannot be used for DTLS RTCP, which was
the reason why RTCP encryption would fail.
ASTERISK-25642
Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a
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