This avoids the following warning:
../plugins/sip/calls-sip-origin.c: In function ‘sip_r_register’:
../plugins/sip/calls-sip-origin.c:483:26: warning: the comparison will always evaluate as ‘true’ for the address of ‘m_url’ will never be NULL [-Waddress]
483 | if (sip->sip_contact && sip->sip_contact->m_url && sip->sip_contact->m_url->url_host) {
| ^~
In file included from /usr/include/sofia-sip-1.12/sofia-sip/nua.h:47,
from ../plugins/sip/calls-sip-util.h:28,
from ../plugins/sip/calls-sip-call.h:30,
from ../plugins/sip/calls-sip-origin.c:31:
/usr/include/sofia-sip-1.12/sofia-sip/sip.h:477:23: note: ‘m_url’ declared here
477 | url_t m_url[1]; /**< SIP URL */
| ^~~~~
../plugins/sip/calls-sip-origin.c: In function ‘sip_callback’:
../plugins/sip/calls-sip-origin.c:779:23: warning: the comparison will always evaluate as ‘true’ for the address of ‘a_url’ will never be NULL [-Waddress]
779 | if (sip->sip_from && sip->sip_from->a_url &&
| ^~
/usr/include/sofia-sip-1.12/sofia-sip/sip.h:386:22: note: ‘a_url’ declared here
386 | url_t a_url[1]; /**< URL */
| ^~~~~
This enables proper negotiation of the codec when answering calls, which
previously also responded with codecs that were not part of the users
preferred ones.
Fixes: #413
A property of type SipMediaEncryption is added to both the origin and
the call which allows to state if we want the media session to be
encrypted with SRTP.
Logic is added to interact with the CallsSdpCryptoContext if encryption
is desired.
Objects of this type keep track of SDP of the local and remote peers,
allow generating offers and answers and codify default policy used for
cryptographic parameters.
Allows setting up cryptographic parameters with
calls_sip_media_pipeline_set_crypto() and use them to set GstCaps for
GstSrtpDec and GObject properties for GstSrtpEnc
by adding functions to the public API which determine if state changes
should be shown to the user and use them (instead of duplicating similar
logic).
We only have a single source of settings, so we should reflect that by
using a singleton. This also reduces our LoC.
This doesn't impair our ability to run tests because there we run with
GSETTINGS_BACKEND=memory
The rtpbin will request GstSrtpDec and GstSrtpEnc elements using the
"request-{rtp,rtcp}-{de,en}coder" family of signals.
The newly added boolean use_srtp controls whether the srtp elements are
returned in the signal handler and thus decides if SRTP is used or not.
Ust GST_DEBUG_BIN_TO_DOT_FILE to generate a dot graph of a pipeline for
debugging purposes when SIGUSR2 is received.
Note the same signal is also used within the dummy plugin to simulate an
incoming call from an unknown number, so when testing you probably want either
the sip plugin or the dummy plugin, but not both.
We were using two distinct pipelines, one for receiving and one for
sending. The receive pipeline was set to the playing state to allocate
the sockets which we would reuse for the sending direction for our NAT
traversal scheme.
The rework to a single pipeline broke reusing sockets subtly.
This happened because the state of the GstUDPSrc could be reset leading
to newly allocated sockets once the pipeline is set to play.
This is now fixed by locking the state of the GstUDPSrc in the ready
state during socket reuse setup and while the pipeline is paused.
Additionally get rid of the "close-socket" property on the udp sources
because it was never needed.
Fixes aa446f82
sq
Using a single pipeline makes implementing encryption easier because we don't
need to duplicate srtpenc and srtpdec elements for each direction.
It also makes it easier to switch to using farstream down the line (see #426).
If the environment variable GST_DEBUG_DUMP_DOT_DIR is set, a graph of the send
and receive pipelines will be written to disk.
To generate a png from the exported dot files graphviz can be used like this:
`dot -Tpng -oimage.png graph.dot`
Now that initialization is split per pipeline and that the OS handles port
allocation we can move setting up socket reuse into the pipeline initialization
step instead of setting it up when starting the media pipelines.
This makes the calls_sip_media_pipeline_start() method a bit simpler.
We're also now reusing sockets for RTCP.
Closes#315
We're not setting the desired ports from the outside anymore, but rather
querying the ports that have been allocated by the operating system.
Therefore the lport-rtp and lport-rtcp property have become superfluous and are
being removed. We also adapt to changes outside of the pipeline code.
We don't expect the initialization to be able to fail. The only thing that could
potentially fail is setting up codecs and this has been delayed until after
initialization.
First of we get rid of the bindings between from "lport-rtp" and "lport-rtcp" to
the "port" property of the udpsrc elements. The properties themselves will get
removed a little later as the required changes are rather intrusive and we need
some more infrastructure in place before we can do the switch.
