* Pulled finding the rtcp-mux attribute flag out of the ICE candidate for
loop. Also ordered the RTCP ICE candidate skip test to fail earlier.
Change-Id: I8905d9c68563027a46cd3ae14dbcc27e9c814809
This sets up the "plumbing" in bridge_softmix to
be able to accommodate Asterisk asking as an SFU
(selective forwarding unit) for conferences.
The way this works is that whenever a channel enters or leaves a
conference, all participants in the bridge get sent a stream topology
change request. The topologies consist of the channels' original
topology, along with video destination streams corresponding to each
participants' source video streams. So for instance, if Alice, Bob, and
Carol are in the conference, and each supplies one video stream, then
the topologies for each would look like so:
Alice:
Audio,
Source video(Alice),
Destination Video(Bob),
Destination video (Carol)
Bob:
Audio,
Source video(Bob)
Destination Video(Alice),
Destination video (Carol)
Carol:
Audio,
Source video(Carol)
Destination Video(Alice),
Destination video (Bob)
This way, video that arrives from a source video stream can then be
copied out to the destination video streams on the other participants'
channels.
Once the bridge gets told that a topology on a channel has changed, the
bridge constructs a map in order to get the video frames routed to the
proper destination streams. This is done using the bridge channel's
stream_map.
This change is bare-bones with regards to SFU support. Some key features
are missing at this point:
* Stream limits. This commit makes no effort to limit the number of
streams on a specific channel. This means that if there were 50 video
callers in a conference, bridge_softmix will happily send out topology
change requests to every channel in the bridge, requesting 50+
streams.
* Configuration. The plumbing has been added to bridge_softmix, but
there has been nothing added as of yet to app_confbridge to enable SFU
video mode.
* Testing. Some functions included here have unit tests.
However, the functionality as a whole has only been verified by
hand-tracing the code.
* Selectivenss. For a "selective" forwarding unit, this does not
currently have any means of being selective.
* Features. Presumably, someone might wish to only receive video from
specific sources. There are no external-facing functions at the moment
that allow for users to select who they receive video from.
* Efficiency. The current scheme treats all video streams as being
unidirectional. We could be re-using a source video stream as a
desetnation, too. But to simplify things on this first round, I did it
this way.
Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d
* Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream()
handle generating disabled/declined streams.
* Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not
check the offerer side negotiated SDP because that isn't the purpose of
this patch and there is much to be done to handle declined/dummy streams.
* Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and
/main/sdp/sdp_merge_crisscross unit tests.
Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31
* Tried to give better variable names.
* Made our SDP answer use the offer's RTP payload types as the SDP RFC
says we SHOULD.
* Updating the local topology now takes the stream format caps. We are
likely preparing to send an offer.
Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0
The sdp_state.remote_capabilities was only used inside merge_sdps() and
subsequent calls to merge_sdps() by re-INVITE's would leak them.
Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce
When we optionally set the interface_address we are forcing the media to
go out a specific interface address. This allows us to optionally have
the media go out the interface that SIP signalling came in on or if we are
configured to have the media always go out a specific address.
Change-Id: I160d9fac322a075bd2557b430632544178196189
The telephone_event option was used as a flag and a bit mapped value in
different places when it is a boolean. It is also inadequate to configure
the DTMF operation of the RTP instance created for the stream.
Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b
RFC 5576 defines how SSRC-level attributes may be added to SDP media
descriptions. In general, this is useful for grouping related SSRCes,
indicating SSRC-level format attributes, and resolving collisions in RTP
SSRC values. These attributes are used widely by browsers during WebRTC
communications, including attributes defined by documents outside of RFC
5576.
This commit introduces the addition of SSRC-level attributes into SDPs
generated by Asterisk. Since Asterisk does not tend to use multiple
SSRCs on a media stream, the initial support is minimal. Asterisk
includes an SSRC-level CNAME attribute if configured to do so. This at
least gives browsers (and possibly others) the ability to resolve SSRC
collisions at offer-answer time.
In order to facilitate this, the RTP engine API has been enhanced to be
able to retrieve the SSRC and CNAME on a given RTP instance.
res_rtp_asterisk currently does not provide meaningful CNAME values in
its RTCP SDES items, and therefore it currently will always return an
empty string as the CNAME value. A task in the near future will result
in res_rtp_asterisk generating more meaningful CNAMEs.
Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789
This change adds a T.38 format which can be used in a stream
topology to specify that a UDPTL stream needs to be created.
The SDP API has been changed to understand T.38 and create
the UDPTL session, add the attributes, and parse the attributes.
This change does not change the boundary of the T.38 state
machine. It is still up to the channel driver to implement and
act on it (such as queueing control frames or reacting to them).
ASTERISK-26949
Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7
The gist of this work ensures that when a remote SDP is received, it is
merged properly with the local capabilities. The remote SDP is converted
into a stream topology. That topology is then merged with the current
local topology on the SDP state. That new merged topology is then used
to create an SDP. Finally, adjustments are made to RTP instances based
on knowledge gained from the remote SDP.
There are also a battery of tests in this commit that ensure that some
basic SDP merges work as expected.
While this may not sound like a big change, it has the property that it
caused lots of ancillary changes.
* The remote SDP is no longer stored on the SDP state. Biggest reason:
there's no need for it. The remote SDP is used at the time it is being
set and nowhere else.
* Some new SDP APIs were added in order to find attributes and convert
generic SDP attributes into rtpmap structures.
* Writing tests made me realize that retrieving a value from an SDP
options structure, the SDP options needs to be made const.
* The SDP state machine was essentially gutted by a previous commit.
Initially, I attempted to reinstate it, but I found that as it had
been defined, it was not all that useful. What was more useful was
knowing the role we play in SDP negotiation, so the SDP state machine
has been transformed into an indicator of role.
* Rather than storing separate local and joint stream state
capabilities, it makes more sense to keep track of current stream
state and update it as things change.
Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d
This change cleans up state management for media streams by moving
RTP instances into their own session structure and adding additional
details that are not relevant to the core (such as connection address).
These can live either in the local capabilities or joint capabilities.
The ability to set explicit connection address information for
the purposes of direct media and NAT has also been added at the
global and stream specific level.
ASTERISK-26900
Change-Id: If7e5307239a9534420732de11c451a2705b6b681
* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.
Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48
This introduces and documents the various states in the state machine.
This also introduces API functions that induce state changes, and places
TODO comments telling what needs to be done in addition to what is
already there. Those TODOs will be replaced with real code in upcoming
changes.
Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed
This establishes the basic allocation/destruction of an SDP state
object, plus some of the simpler getter methods involved. Subsequent
tasks will deal with adding a state machine, creating SDPs from
capabilities and options, and merging SDPs into a joint SDP.
Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709
