These are generated at runtime (depending if we are doing Plan-B or
Unified Plan). Currently this field is actually unused, so just getting
rid of it.
Relates to #106
We will implement these eventually, but for now just delete them.
We need to explore how we can automatically figure out what parts
of WebRTC we haven't implemented yet.
Relates to #106
Fix RemoteDescription parsing so that it parses all sections when
looking for a fingerprint, before it would fail if the first section
did not contain one
This only does parsing, does not generate them.
In the future this option will be exposed via the SettingEngine,
including the ability to opt out of mDNS completely for security/scaling
concerns
Relates to pion/webrtc#699
Users are unable to use the callbacks inside `internal/ice`.
Even though we alias things like OnSelectedCandidatePairChange
are unusable since in the package they use `ice.Candidate` instead of
`ICECandidate`
Fix inconsistency with error handling when we have no candidate pairs.
Before we had custom code in RTP handling that would discard errors
if it was because we had no candidate pairs. Move this logic into the
mux so we have consistent behavior with Datachannels
This can be expected and is a soft failure. Every subsystem is expected
to handle lossy communication.
Resolves#706
Moved ice related files to internal/ice. Added ice.go and
ice_go.go files containing aliases to internal/ice. Added
internal/tools/gen/genaliasdocs.go helper script for generating the
documentation for these aliases. Dropped ice
prefix from file names in internal/ice package.
Data channel creation checks if the sctp transport is already
established. Because this check was incorrect, in some cases it returned
an error instead of delaying opening the data channel until the sctp
connection is established. Fixed the check. Added locking to
CreateDataChannel to avoid race condition. Added test.
Before our RTPTransceiver logic would only allow Sendrecv to
match with another Sendrecv. Instead allow a SendRecv to be used
with a local Recvonly
Co-authored-by: Sean DuBois <sean@siobud.com>
Previously the PeerConnection would create invalid SDPs
when answering an offer that contained codecs it didn't
know about. Now it rejects them by setting the media format
to 0.
Related to #449
By default, we should be using unified-plan as the
SDP format of choice. This patch adds a PeerConnection
configuration option to allow the user to specify that
they want plan-b only, unified-plan only, or a special
compatibility mode where a plan-b answer will be generated
IFF a plan-b offer is received.
Instead of printing the error to stdout return the error to the user.
This may not be a hard error (as later certificates would have passed)
but it never is good to be in a state where you have certificates in a
broken state.
Resolves#586
This commit has breaking changes. This API change means we
can no longer support an arbitrary number of receivers. For every track
you want to receive you MUST call PeerConnection.AddTransceiver
We do now support sending an multiple audio/video feeds. You can see
this behavior via gstreamer-receive and gstreamer-send currently.
Resolves#54
Writing to a Track shouldn't return errors for an individual
RTPSender. This filters ErrNoCandidatePairs from being returned
and instead just returns nil
Resolves#523
Add OnICECandidate and OnICEGatheringStateChange methods to
PeerConnection. The main goal of this change is to improve API
compatibility with the JavaScript/Wasm bindings. It does not actually
add trickle ICE support or change the ICE candidate gathering process,
which is still synchronous in the Go implementation. Rather, it fires
the appropriate events similar to they way they would be fired in a true
trickle ICE process.
Remove unused OnNegotiationNeeded event handler. This handler is not
required for most applications and would be difficult to implement in
Go. This commit removes the handler from the JavaScript/Wasm bindings,
which leads to a more similar API for Go and JavaScript/Wasm.
Add OnICEGatheringStateChange to the JavaScript/Wasm bindings. Also
changes the Go implementation so that the function signatures match.
Syntax could be better but works in a standards-compliant way.
```
codec := webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000)
codec.RTPCodecCapability.RTCPFeedback = []webrtc.RTCPFeedback{{
Type: "nack",
}, {
Type: "nack",
Parameter: "pli",
}}
```
- Fix an issue with ipv6 srflx candidates gathering.
- Add SetNetworkTypes config to SettingEngine to control what network
types are allowed to be connected.
Resolves#460
Resolves#478. Adds minimal JavaScript/WASM bindings. This makes it
possible to compile core parts of pions/webrtc to WASM and run it in the
browser. Only data channels are supported for now and there is
limited/no support for certificates and credentials.
DTLS shutdown deadlocks if Close is called before startup completes,
because the DTLS connection hasn't finished yet we don't have handles
to close anything.
This updates DTLS to follow how SCTP is shutdown, by shutting down the
nextConn (ICE in this case) we can shutdown the subsystem. By closing
ICE first, DTLS (and then SCTP) close properly no matter what state
they are in.
Resolves#487
Mostly for PeerConnection.AddTrack RTPTransceiver filtering.
Also, removed nil-check for RTPSender.track:
1) RTPSender is always instantiated with PeerConnection.NewRTPSender
2) PeerConnection.NewRTPSender returns error if nil track passed
