[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",
]