rtp-core 0.3.0

RTP media transport with jitter buffer and codec support
Documentation
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[package]
edition = "2021"
name = "rtp-core"
version = "0.3.0"
build = false
autolib = false
autobins = false
autoexamples = false
autotests = false
autobenches = false
description = "RTP media transport with jitter buffer and codec support"
homepage = "https://github.com/xmppjingle/sipr"
readme = false
keywords = [
    "rtp",
    "audio",
    "voip",
    "jitter-buffer",
    "g711",
]
categories = [
    "multimedia::audio",
    "network-programming",
]
license = "Apache-2.0"
repository = "https://github.com/xmppjingle/sipr"

[features]
audio-device = ["cpal"]
default = [
    "audio-device",
    "opus",
]
opus = ["audiopus"]

[lib]
name = "rtp_core"
path = "src/lib.rs"

[dependencies.audiopus]
version = "0.3.0-rc.0"
optional = true

[dependencies.bytes]
version = "1"

[dependencies.cpal]
version = "0.15"
optional = true

[dependencies.rand]
version = "0.8"

[dependencies.serde]
version = "1"
features = ["derive"]

[dependencies.thiserror]
version = "1"

[dependencies.tokio]
version = "1"
features = ["full"]

[dependencies.tracing]
version = "0.1"

[dev-dependencies.tokio]
version = "1"
features = [
    "full",
    "test-util",
]