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# webrtc-data changelog | ||
# rtc-datachannel changelog | ||
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## Unreleased | ||
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* Remove builder pattern from `data_channel::Config` [#411](https://github.com/webrtc-rs/webrtc/pull/411). | ||
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## v0.7.0 | ||
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* Increased required `webrtc-sctp` version to `0.8.0`. | ||
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## v0.6.0 | ||
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* Increased minimum support rust version to `1.60.0`. | ||
* Do not loose data in `PollDataChannel::poll_write` [#341](https://github.com/webrtc-rs/webrtc/pull/341). | ||
* `PollDataChannel::poll_shutdown`: make sure to flush any writes before shutting down [#340](https://github.com/webrtc-rs/webrtc/pull/340) | ||
* Increased required `webrtc-util` version to `0.7.0`. | ||
* Increased required `webrtc-sctp` version to `0.7.0`. | ||
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### Breaking changes | ||
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* Make `DataChannel::on_buffered_amount_low` function non-async [#338](https://github.com/webrtc-rs/webrtc/pull/338). | ||
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## v0.5.0 | ||
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* [#16 [PollDataChannel] reset shutdown_fut future after done](https://github.com/webrtc-rs/data/pull/16) by [@melekes](https://github.com/melekes). | ||
* Increase min verison of `log` dependency to `0.4.16`. [#250 Fix log at ^0.4.16 to make tests compile](https://github.com/webrtc-rs/webrtc/pull/250) by [@k0nserv](https://github.com/k0nserv). | ||
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## Prior to 0.4.0 | ||
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Before 0.4.0 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/data/releases). |
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# webrtc-dtls changelog | ||
# rtc-dtls changelog | ||
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## Unreleased | ||
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## v0.7.1 | ||
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* Added support for insecure/deprecated signature verification algorithms [#342](https://github.com/webrtc-rs/webrtc/pull/342) by [@chuigda](https://github.com/chuigda). | ||
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## v0.7.0 | ||
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* Increased minimum support rust version to `1.60.0`. | ||
* Add `RTCCertificate::from_pem` and `RTCCertificate::serialize_pem` (only work with `pem` feature enabled) [#333](https://github.com/webrtc-rs/webrtc/pull/333) | ||
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### Breaking | ||
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* Increased required `webrtc-util` version to `0.7.0`, with this change some methods in `DTLSConn` that implement `webrtc_util::Conn` have changed from async to sync. | ||
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## v0.6.0 | ||
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* [#254 [DTLS] Add NamedCurve::P384](https://github.com/webrtc-rs/webrtc/pull/254) contributed by [neonphog](https://github.com/neonphog) | ||
* Increased min verison of `log` dependency to `0.4.16`. [#250 Fix log at ^0.4.16 to make tests compile](https://github.com/webrtc-rs/webrtc/pull/250) by [@k0nserv](https://github.com/k0nserv). | ||
* Increased serde's minimum version to 1.0.110 [#243 Fixes for cargo minimal-versions](https://github.com/webrtc-rs/webrtc/pull/243) contributed by [algesten](https://github.com/algesten) | ||
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## Prior to 0.6.0 | ||
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Before 0.6.0 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/dtls/releases). |
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# webrtc-ice changelog | ||
# rtc-ice changelog | ||
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## Unreleased | ||
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### Breaking changes | ||
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* remove non used `MulticastDnsMode::Unspecified` variant [#404](https://github.com/webrtc-rs/webrtc/pull/404): | ||
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## v0.9.0 | ||
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* Increased minimum support rust version to `1.60.0`. | ||
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### Breaking changes | ||
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* Make functions non-async [#338](https://github.com/webrtc-rs/webrtc/pull/338): | ||
- `Agent`: | ||
- `get_bytes_received`; | ||
- `get_bytes_sent`; | ||
- `on_connection_state_change`; | ||
- `on_selected_candidate_pair_change`; | ||
- `on_candidate`; | ||
- `add_remote_candidate`; | ||
- `gather_candidates`. | ||
- `unmarshal_candidate`; | ||
- `CandidateHostConfig::new_candidate_host`; | ||
- `CandidatePeerReflexiveConfig::new_candidate_peer_reflexive`; | ||
- `CandidateRelayConfig::new_candidate_relay`; | ||
- `CandidateServerReflexiveConfig::new_candidate_server_reflexive`; | ||
- `Candidate`: | ||
- `addr`; | ||
- `set_ip`. | ||
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## v0.8.2 | ||
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* Add IP filter to ICE `AgentConfig` [#306](https://github.com/webrtc-rs/webrtc/pull/306) and [#318](https://github.com/webrtc-rs/webrtc/pull/318). | ||
* Add `rust-version` at 1.57.0 to `Cargo.toml`. This was already the minimum version so does not constitute a change. | ||
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## v0.8.1 | ||
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This release was released in error and contains no changes from 0.8.0. | ||
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## v0.8.0 | ||
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* Increased min verison of `log` dependency to `0.4.16`. [#250 Fix log at ^0.4.16 to make tests compile](https://github.com/webrtc-rs/webrtc/pull/250) by [@k0nserv](https://github.com/k0nserv). | ||
* Incresed serde's minimum version to 1.0.102 [#243 Fixes for cargo minimal-versions](https://github.com/webrtc-rs/webrtc/pull/243) contributed by [algesten](https://github.com/algesten) | ||
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## Prior to 0.8.0 | ||
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Before 0.8.0 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/ice/releases). |
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# rtcp changelog | ||
# rtc-rtcp changelog | ||
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## Unreleased | ||
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## v0.7.2 | ||
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* [#372 Fix over-NACK due not resetting lost_packets bitmask](https://github.com/webrtc-rs/webrtc/pull/372/) | ||
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## v0.7.1 | ||
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* Increased minimum support rust version to `1.60.0`. | ||
* Increased required `webrtc-util` version to `0.7.0`. | ||
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## v0.7.0 | ||
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* [#14 Prevent crash in RTCP NACK writing](https://github.com/webrtc-rs/rtcp/pull/14) by [@pthatcher](https://github.com/pthatcher). | ||
* Adds `IntoIterator` for `NackPair` which iterates over all the sequence numbers specified by the `NackPair`. This is similar to `packet_list` but without requiring the allocation of a Vec. Added in [#225 Add RTP Stats to stats report](https://github.com/webrtc-rs/webrtc/pull/225) by [@k0nserv](https://github.com/k0nserv). | ||
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## Prior to 0.7.0 | ||
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Before 0.7.0 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/rtcp/releases). | ||
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# rtp changelog | ||
# rtc-rtp changelog | ||
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## Unreleased | ||
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## v0.6.8 | ||
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* Increased minimum support rust version to `1.60.0`. | ||
* Adds a new generic header extensions type `rtp::extension::HeaderExtension` which allows abstracting over all known extensions as well as custom extensions. [#336](https://github.com/webrtc-rs/webrtc/pull/336) by [@k0nserv](https://github.com/k0nserv). | ||
* Added video orientation(`urn:3gpp:video-orientation`) extension support. [#331](https://github.com/webrtc-rs/webrtc/pull/331) by [@algesten](https://github.com/algesten). | ||
* Allow RTP extensions to be serialized and deserialized via serder. [#332](https://github.com/webrtc-rs/webrtc/pull/332) by [@algesten](https://github.com/algesten). | ||
* Increased required `webrtc-util` version to `0.7.0`. | ||
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## v0.6.7 | ||
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* Bumped util dependecy to `0.6.0`. | ||
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## Prior to 0.6.7 | ||
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Before 0.6.7 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/rtp/releases). | ||
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# rtc-sctp changelog | ||
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## Unreleased |
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