webrtc
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webrtc-rs
A pure Rust implementation of WebRTC
Regarding the Go version plan: https://github-wiki-see.page/m/pion/webrtc/wiki/Goals-2021 **Add pion/interceptor jitter_buffer implementation Cache incoming packets and re-arrange them Allow users to define cache behavior (% loss, delay etc..)** I would love to...
👋 Hi! I've been working a bit on a SFU example to add to the examples in https://github.com/webrtc-rs/examples/pull/15. This is a good starting point for what I think is the...
When writing futures that are likely to run in race with others using e.g. `tokio::select!` it's important to consider cancel safety. A typical pattern that isn't cancel safe is ```rust...
Pion added av1 support: https://github.com/pion/webrtc/pull/2071 https://github.com/pion/webrtc/pull/2176
### Discussed in https://github.com/webrtc-rs/webrtc/discussions/140 Originally posted by **yusufziyacs** December 11, 2021 Hello there. I've just started using webrtc-rs in order to convert RTSP streams into webrtc. I can easily create...
hi This is an issue reopen from 174, as state before the code below will leave some socket unreleased in the task/thread/process ``` let pc = peer_connection.clone(); peer_connection .on_peer_connection_state_change(Box::new(move |s:...
The test case is using Chrome to webrtc-rs. 1. Add mic track in Chrome. 2. Remove mic track. 3. Re-add mic track. The resulting SDP dance is here: ``` Chrome...
I see that the [stats module](https://github.com/webrtc-rs/webrtc/blob/master/src/stats/mod.rs) exists, but does not have any content. I was hoping to contribute a pull request adding RTP stats, and wanted to make sure that...
I'm getting `Rtp(ErrShortPacket)` after several successful depacketize calls from the `H264Packete` depacketizer, and am struggling to understand what I'm doing wrong. The minimal reproduction of this is: ```rust use std::sync::Arc;...
https://blog.rust-lang.org/2021/10/21/Rust-1.56.0.html