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WebRTC.rs

WebRTC in Rust. A pure Rust implementation of WebRTC stack.
Collective - Host: opensource - https://opencollective.com/webrtc-rs - Website: https://webrtc.rs - Code: https://github.com/webrtc-rs

add track_remote_rx await

github.com/webrtc-rs/sfu - cc726cf459216a38dc44e3c9a0b952dba8fc1bfd authored 11 months ago by yngrtc <[email protected]>
add data_channel_rx await

github.com/webrtc-rs/sfu - bf893bc6322be595da2978ac91399a798db92cfe authored 11 months ago by yngrtc <[email protected]>
update rtc pointer

github.com/webrtc-rs/sfu - 88fd1c66f8a5f4140e61c15e87d677b8a102affc authored 11 months ago by yngrtc <[email protected]>
add test_play_from_disk_vpx_1to1

github.com/webrtc-rs/sfu - 94ffec627c1f3b4148cdce23e19ffb3124cde234 authored 11 months ago by yngrtc <[email protected]>
use Docker to run integration tests

github.com/webrtc-rs/sfu - 7dba3c4e1fc5d70be2067aa4118f7fa6436ee05d authored 11 months ago by yngrtc <[email protected]>
implement test_data_channels

github.com/webrtc-rs/sfu - 6fc0cdce1b38fa505b85859adb9b1b72f3f677a1 authored 11 months ago by yngrtc <[email protected]>
implement test_datachannel

github.com/webrtc-rs/sfu - 66a2bbee3f84c543f24ec45265a4e2842d5aa693 authored 11 months ago by yngrtc <[email protected]>
aadd webrtc/webrtc as dev-dep

github.com/webrtc-rs/sfu - ae25e2a9a64a695c66f35b4f2034f76c801d732a authored 11 months ago by yngrtc <[email protected]>
add submodule webrtc

github.com/webrtc-rs/sfu - d865e14edcecb2d6ab1498151f186774bb72e85f authored 11 months ago by yngrtc <[email protected]>
setup integration tests

github.com/webrtc-rs/sfu - 40679fc06be733ab1658e624943b314f7568d57c authored 11 months ago by yngrtc <[email protected]>
add integration tests in github actions

github.com/webrtc-rs/sfu - 8efd0838af935d795cea7e11e91449f59a48725d authored 11 months ago by yngrtc <[email protected]>
don't forward RR and TF

github.com/webrtc-rs/sfu - cf3746c49ba6b57842a3ba6ca5192243b2108767 authored 11 months ago by yngrtc <[email protected]>
sdp: Use vector instead of HashMap for SimulcastRid (#532)

This makes sure that rids are not reordered when doing offer <-> answer

github.com/webrtc-rs/webrtc - b71c822d8c0b5a697214009ad1705625e1cc541d authored 11 months ago by Haakon Sporsheim <[email protected]>
update Cargo.lock to fix nightly test due to "curve25519-dalek"

version = "4.1.1" issue, now it is 4.1.2

github.com/webrtc-rs/webrtc - 2d7e56ae2abb686a4096164d8de9d6d38b663c8b authored 11 months ago by Rusty Rain <[email protected]>
add receiver_stream

github.com/webrtc-rs/sfu - c2a2812b88403b029740cf80660eb0d9567b1983 authored 11 months ago by yngrtc <[email protected]>
move interceptor into Endpoint

github.com/webrtc-rs/sfu - 9f2f3eb2b928cccfbba9156b453997cd57f6ee54 authored 11 months ago by yngrtc <[email protected]>
fix sender_report bug

github.com/webrtc-rs/sfu - 3c0b9067523a83e8e9a4bccf474c28c25901fdc3 authored 11 months ago by yngrtc <[email protected]>
complete SenderReport Interceptor

github.com/webrtc-rs/sfu - a7f82b5dae0327cdf14cb81b5b1c7c1f62ff4ac8 authored 11 months ago by yngrtc <[email protected]>
refactor Registry

github.com/webrtc-rs/sfu - 21c5f484643378787c7f3654d1e8dbbce106a88a authored 11 months ago by yngrtc <[email protected]>
add sender/receiver_report framework

github.com/webrtc-rs/sfu - 5d29d0a411c9b56ea7745d74ed12ce6529e5fa98 authored 11 months ago by yngrtc <[email protected]>
add Registry to MediaConfig

github.com/webrtc-rs/sfu - c17fc3236c6a1b7f1e2068137909ad85d79e9307 authored 11 months ago by yngrtc <[email protected]>
use chain of responsibility design pattern for interceptor

github.com/webrtc-rs/sfu - b3149a007cd6b04fa0e09cb63c3fd72a0a1c61e2 authored 11 months ago by yngrtc <[email protected]>
define Interceptor trait

github.com/webrtc-rs/sfu - d0dfd653f16ff7db151278540bbb2f3c136f0cca authored 11 months ago by yngrtc <[email protected]>
merge rtp_handler into srtp_handler

github.com/webrtc-rs/sfu - b2b1fd08ff66233579d624e04822b367ed290d98 authored 11 months ago by Rusty Rain <[email protected]>
temporally disable default RTCP feedback

github.com/webrtc-rs/sfu - b7215032c4795f75f81572a9341aab7f66a1084a authored 11 months ago by Rusty Rain <[email protected]>
change default host to 127.0.0.1 for chat example

github.com/webrtc-rs/sfu - 1664fd90ac102340a04b9ad70a756e4cf9b8c5cd authored 11 months ago by Rusty Rain <[email protected]>
use server_config's media_config for SDP popluate

github.com/webrtc-rs/sfu - a3ea275487c4b7421d306887052d0e36a205b239 authored 11 months ago by Rusty Rain <[email protected]>
add MediaConfig to ServerConfig

github.com/webrtc-rs/sfu - cc70209af43aa3068004f21a48f035ac7d81a2db authored 11 months ago by Rusty Rain <[email protected]>
rename MediaEngineHeaderExtension to RTCRtpHeaderExtension

github.com/webrtc-rs/sfu - 313a4b266dfe04dc21f557d2791e8164573050c0 authored 11 months ago by Rusty Rain <[email protected]>
add media_config to register codecs/feedbacks/header_extensions #2

github.com/webrtc-rs/sfu - 3eab2b30fec12fb9e628808370bfb32663e3ce1c authored 11 months ago by Rusty Rain <[email protected]>
rename handlers to handler

github.com/webrtc-rs/sfu - a285f2f60ce7f5cfac2e06f30fb9de65bde28b10 authored 11 months ago by Rusty Rain <[email protected]>
move session and endpoint to top

github.com/webrtc-rs/sfu - f5c0c2956730c978c14e21f72a584650eb27b78c authored 11 months ago by Rusty Rain <[email protected]>
move description to top

github.com/webrtc-rs/sfu - d5021f9755413c241e0bd41029a6a13c2e052f18 authored 11 months ago by Rusty Rain <[email protected]>
fix clippy warning

github.com/webrtc-rs/sfu - b8fa4603aea681ba03d4444079068ad551787de3 authored 11 months ago by Rusty Rain <[email protected]>
refactor public use

github.com/webrtc-rs/sfu - d05d6392be770a507447e9ba22f9dea8ec3b1782 authored 11 months ago by Rusty Rain <[email protected]>
use SctpConfig to config sctp-port and max-message-size for SDP #4

github.com/webrtc-rs/sfu - 1f9b1ea79e5ae015e3996d5656e9f18f1f1436fe authored 11 months ago by Rusty Rain <[email protected]>
use builder pattern for ServerConfig

github.com/webrtc-rs/sfu - 198b900e81f48b898cb8b7fc339505634bb0f5ef authored 11 months ago by Rusty Rain <[email protected]>
update rtc pointer

github.com/webrtc-rs/sfu - c8bca5dd0e1e6dae1d85558fb3ad48b5d31c3347 authored 11 months ago by Rusty Rain <[email protected]>
update rtc pointer

github.com/webrtc-rs/sfu - 837fbe15124b85b4ed5a516de68c355f73fe8fee authored 11 months ago by yngrtc <[email protected]>
bump version to v0.1.7

github.com/webrtc-rs/sctp-proto - d154eb0c227c8401a9410e4ab5ce332edafceee7 authored 11 months ago by yngrtc <[email protected]>
Merge pull request #7 from davibe/fix/T3RTX-starvation

Fix T3RTX timer starvation

github.com/webrtc-rs/sctp-proto - fbefad6711569e95a4dc390cdde557ed83ff1f18 authored 11 months ago by Rusty Rain <[email protected]>
Fix T3RTX timer starvation

When recovering from temporary network loss, regular traffic can result
in repeated acks, which ...

github.com/webrtc-rs/sctp-proto - 42aba09c18de145d364777179fcfd76bfb8c1ba6 authored 11 months ago by Davide Bertola <[email protected]>
fix duplicated mid bug in handle_datachannel_open

github.com/webrtc-rs/sfu - c918dcd032e1c9079ea89ccde3822ae9733a915d authored 11 months ago by yngrtc <[email protected]>
refactor logging

github.com/webrtc-rs/sfu - 358a0849b57083718c87f5f26244bb7cc8badf51 authored 11 months ago by yngrtc <[email protected]>
only create offer message when SDP renegotiation is needed

github.com/webrtc-rs/sfu - 85b9bfbf872818cd509b2aa17be3c509f1fb2ce2 authored 11 months ago by yngrtc <[email protected]>
remove Rc/RefCell inside ServerStates/Session/Endpoint

github.com/webrtc-rs/sfu - acba1ef83f187110ff682f10c54a84dc2869619b authored 11 months ago by yngrtc <[email protected]>
fix mid order

github.com/webrtc-rs/sfu - d9efad8d28bc48fcc2be99b7ba2fa6978f421c19 authored 11 months ago by yngrtc <[email protected]>
refactor rtp_transceiver

github.com/webrtc-rs/sfu - dff0bbe6dbeb68dffb7227101fcfbfcaf2cd6faa authored 11 months ago by yngrtc <[email protected]>
add accept_answer

github.com/webrtc-rs/sfu - 3c9465e64b0bfd8c26bcb603b6947b58d08b224e authored 11 months ago by yngrtc <[email protected]>
add forward other offers

github.com/webrtc-rs/sfu - 1c3501e4778c0a088527abf8c333bd76eb807318 authored 11 months ago by yngrtc <[email protected]>
add set_association_handle_and_stream_id in Transport

github.com/webrtc-rs/sfu - f440aa55884dc9d43429a1bbfc67b95a093c8477 authored 11 months ago by yngrtc <[email protected]>
add create_offer

github.com/webrtc-rs/sfu - 6a71d1c14de14d0abee7de965aa4e94e8cdc5954 authored 11 months ago by yngrtc <[email protected]>
change payload in ApplicationMessage to DataChannelEvent with Open/Message/Close

github.com/webrtc-rs/sfu - 7ad853b4687eaa3cc54b2cf3bf1e1a53eba076fc authored 11 months ago by yngrtc <[email protected]>
refactoring rtp_transceiver

github.com/webrtc-rs/sfu - c791d726befe203ca470d3391b1308299ce4123c authored 11 months ago by yngrtc <[email protected]>
impl handle_rtp_message/handle_rtcp_message

github.com/webrtc-rs/sfu - 21665e01ffa07595fdd5b73a85f52b829bc0499b authored 11 months ago by yngrtc <[email protected]>
make transceivers in Endpoint private

github.com/webrtc-rs/sfu - e78e805fb700ea243aa776fdabade94f9c3b06a0 authored 11 months ago by yngrtc <[email protected]>
refactor Transport

github.com/webrtc-rs/sfu - 0304a1e3fd9552e4cc2afe54a71b0d080b3526c1 authored 11 months ago by yngrtc <[email protected]>
add handle_rtp_message/handle_rtcp_message (WIP)

github.com/webrtc-rs/sfu - 987465f3eee05534314c4fec15a93b0ac4262cac authored 12 months ago by yngrtc <[email protected]>
add rtp/rtcp_handler

github.com/webrtc-rs/sfu - 5952b2768bd87ec368a7196436dc62821705dffe authored 12 months ago by yngrtc <[email protected]>
add srtp and move srtp context to endpoint's transport

github.com/webrtc-rs/sfu - ee8bc1dce0148ffda9d61049b133aa0326a93990 authored 12 months ago by yngrtc <[email protected]>
fix ice state disconnected bug

github.com/webrtc-rs/sfu - 272821573a9e729646e1b8cae61e46031558f625 authored 12 months ago by yngrtc <[email protected]>
add console log for chat.html

github.com/webrtc-rs/sfu - f8862efdc81dc9a7efc75e53a4560b4c9cc12681 authored 12 months ago by yngrtc <[email protected]>
Create FUNDING.yml

github.com/webrtc-rs/sfu - a2aae0f448c05266e9273a448c8b094f305a6232 authored 12 months ago by Rusty Rain <[email protected]>
add rtp header extensions support

github.com/webrtc-rs/sfu - bea1789098b09d9c8fcbe0be6f5c24bfb8d02a33 authored 12 months ago by yngrtc <[email protected]>
add sdp util functions

github.com/webrtc-rs/sfu - fdd6d5b6a607b3252e2ceaac2099671c6ec3cb26 authored 12 months ago by yngrtc <[email protected]>
refactor fmtp

github.com/webrtc-rs/sfu - f5bbea756a339d748b0dc0bf9622dfa12827bdeb authored 12 months ago by yngrtc <[email protected]>
fix rtp_transceiver get_codecs (WIP)

github.com/webrtc-rs/sfu - 373b6f8cb81e2901bf3928eabd0934b754b47ddf authored 12 months ago by yngrtc <[email protected]>
add set_remote_description

github.com/webrtc-rs/sfu - e4af34085ada578c5a240b55a87cb69ef4e9e608 authored 12 months ago by yngrtc <[email protected]>
refactor generate_matched_sdp

github.com/webrtc-rs/sfu - 6163a15982868cd214da8c7ee199685b1014b9a4 authored 12 months ago by yngrtc <[email protected]>
refine accept_offer

github.com/webrtc-rs/sfu - 3999ceeec8d1c54d5e5d0d7f9a6f232fb30b674f authored 12 months ago by yngrtc <[email protected]>
add exception_handler

github.com/webrtc-rs/sfu - cfe4d186f096d319a7c519cbbc1bb3a157e7b1b2 authored 12 months ago by yngrtc <[email protected]>
fix data_message_type to Text to be compatible with Chrome

github.com/webrtc-rs/sfu - 6df59f1629370b4c664c8f9cff4ca558df9d2f36 authored 12 months ago by yngrtc <[email protected]>
make sctp internal_buffer max_message_size

github.com/webrtc-rs/sfu - 165217e05806da14728188adef44df2aeef21918 authored 12 months ago by yngrtc <[email protected]>
add handle_dtls_message

github.com/webrtc-rs/sfu - 20e14e6a8b7e465a1dcb065274bdef8721593eb9 authored 12 months ago by yngrtc <[email protected]>
add data channel handler

github.com/webrtc-rs/sfu - 8f466dba65f273817a227b8ee61931db4ce0fd52 authored 12 months ago by yngrtc <[email protected]>
add sctp handler

github.com/webrtc-rs/sfu - 410a9968fd4af02228e458c8050f7223d62d7981 authored 12 months ago by yngrtc <[email protected]>
add dtls handler

github.com/webrtc-rs/sfu - f9fc4890f4b4e5c3d9c4089ccb7f437c3d3f38ea authored 12 months ago by yngrtc <[email protected]>
refactor session_config

github.com/webrtc-rs/sfu - 08eaa5cdda891272ac20062f1a10f3224df7a1b3 authored 12 months ago by yngrtc <[email protected]>
move accept_pending_offer/candidates to server_states

github.com/webrtc-rs/sfu - 25cd7d3c3ca32f4439a292796893ea283719a8f1 authored 12 months ago by yngrtc <[email protected]>
refactor messages/types

github.com/webrtc-rs/sfu - 14194008a3ba853b8badc9683971285dc6d5a598 authored 12 months ago by yngrtc <[email protected]>
add Gateway handler (WIP)

github.com/webrtc-rs/sfu - 2c9540ef31b3e8ee39dd8cf0785fe30feb29cfd9 authored 12 months ago by yngrtc <[email protected]>
add Stun handler

github.com/webrtc-rs/sfu - 22f0ca76170f4de4dc4f4595814363a071a83335 authored 12 months ago by yngrtc <[email protected]>
add Demuxer handler

github.com/webrtc-rs/sfu - 95f588725e1f0a61b9a8471380e1987b2627f7fd authored 12 months ago by yngrtc <[email protected]>
remove generate_unmatched_sdp which is never used in SFU

github.com/webrtc-rs/sfu - 83365455db3cd9919a944b9165d24fb59aae3ce0 authored 12 months ago by yngrtc <[email protected]>
refactoring MediaSection

github.com/webrtc-rs/sfu - 6d499fc641498c0d9b3cb56a07f8d9b14c255d4d authored 12 months ago by yngrtc <[email protected]>
refactoring Candidate

github.com/webrtc-rs/sfu - 5117761a92cebcd5d4bffd4b5ccd3c53784ed233 authored 12 months ago by yngrtc <[email protected]>
refactor RTCRtpSender with negotiated: RefCell<bool>,

github.com/webrtc-rs/sfu - fab3fcdaddf7dace5a132e6bc87296b810d3d147 authored 12 months ago by yngrtc <[email protected]>
refactor code and add generate_unmatched_sdp

github.com/webrtc-rs/sfu - d0ddf5b1064e898d7d035c0098bda64542a107b1 authored 12 months ago by yngrtc <[email protected]>
sdp: simulcast: Properly parse and set 'a=simulcast:' (#528)

This improves parsing and generation of SDP for simulcast attribute
according to RFC8853.
Supp...

github.com/webrtc-rs/webrtc - 0564ff33d88c667e8450c6c4a7487ea5be8440ef authored 12 months ago by Haakon Sporsheim <[email protected]>