Fix an inverted condition when checking if sink pad caps match
the codec-preference of an unassociated transceiver, and
fix a condition check for transceiver media kind to
avoid matching sinkpad requests where caps aren't provided
against unassociated transceivers where the caps might
not match later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7237>
With MR 7156, transceivers and transports are created earlier,
but for sendrecv media we could get `not-linked` errors due to
transportreceivebin not being connected to rtpbin yet when incoming
data arrives.
This condition wasn't being tested in elements_webrtcbin, but could be
reproduced in the webrtcbidirectional example. This commit now also
adds a test for this, so that this doesn't regress anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7294>
According to https://w3c.github.io/webrtc-pc/#set-the-session-description
(steps in 4.6.10.), we should be creating and associating transceivers when
setting session descriptions.
Before this commit, webrtcbin deviated from the spec:
1. Transceivers from sink pads where created when the sink pad was
requested, but not associated after setting local description, only
when signaling is STABLE.
2. Transceivers from remote offers were not created after applying the
the remote description, only when the answer is created, and were then
only associated once signaling is STABLE.
This commit makes webrtcbin follow the spec more closely with regards to
timing of transceivers creation and association.
A unit test is added, checking that the transceivers are created and
associated after every session description is set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7156>
Don't add an extra ref if non-floating as that ref will never be
unreffed.
gst_bin_add() is transfer floating (alias to transfer none).
Fixes a leak when a non-floating ref was provided as a return value in
the request-aux-sender signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6807>
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.
This commit adds all the `ssrc-` attributes from the matching PT entries.
The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.
The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
In rtpbin we already systematically check for all property names
except latency, correct that.
In webrtcbin we need to check before trying to use the do-retransmission
property.
This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
Remove optional sprop-stereo and sprop-maxcapture fields from Opus
remote offer caps before intersecting with local codec preferences.
According to https://datatracker.ietf.org/doc/html/rfc7587#section-7.1
those fields are sender-only informative, and don't affect
interoperability.
Fixes cases where the webrtc media will end up receive-only if the
local side wants to send stereo but the remote is sending mono, or
vice versa.
There may be other fields in other codecs, so the implementation
anticipates needing to add further fields and codecs in the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5993>
The code seems to validate that the media-level fingerprint matches
the fingerprint of the previous media or of the whole session. There
is no such requirement in any RFC I found. The session-session one
is just meant to act as a fallback when there is no media-level
fingerprint.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1118>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
A blocking pad probe is added on new sink pads, it's usually removed after the
caps have been negotiated or the signaling state switched to stable, but if that
never happens and the pad is released we kept the pad probe active, leaving the
pad blocked, preventing clean disposal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4529>
Current implementation can in some cases detect
that all data is sent but in reality it is not,
leading to a push to an unlinked pad.
This is a race between the probe used to track data sent and a
call to close.
This patch sends an EOS before starting the close procedure
and then waits for the EOS event to come through to the
src pad before commencing with tear down.
This ensures that any queued data before EOS is flushed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4462>
`webrtc->signaling_state` (from) and `new_signaling_state` (to) had the
same value when printed in a trace log. This commit adds a
`old_signaling_state` variable to hold the previous value, so that the
print statement works as intented.
Spotted by: Mustafa Asım REYHAN
Fixes#1802
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4362>
In webrtc_data_channel_send functions, both data and string,
an early return on a non-open datachannel caused it to leak
the buffer used for pushing to appsrc, meaning any buffer
sent after leaving the open state was leaked in full.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4191>
The `add_candidate` vfunc of the GstWebRTCICE interface gained a GstPromise
argument, which is an ABI break. We're not aware of any external user of this
interface yet so we think it's OK.
This change is useful in cases where the application needs to bubble up errors
from the underlying ICE agent, for instance when the agent was given an invalid
ICE candidate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3960>
The signal triggers an asynchronous task on the PC thread but in some cases it
can be useful for apps to be notified when the task completed. This method of
the PeerConnection spec also returns a Promise so the interface is now more
coherent with the spec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3960>
The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.
Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.
Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
When picking an available payload type, we need to pick one that is
available across all media.
The previous code, when multiple media were present, looked at the first one,
noticed it had pt 96 as the media pt, then simply looked at the next media,
noticed it didn't, and decided 96 was available.
Instead, check if the pt is used by any of the media, if it is, decide
it is not available and go to the next pt. I'm fairly sure that was the
original intent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2984>
This avoids getting in a bunch of corner cases. We'd have to insert
a "rejected" line from the start as a place-holder to get around this,
but the rest of the code just becomes more complicated, so just
disallow it for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
