Add support for logging RTP/RTCP packets
This commit is contained in:
Jasper Hugo
parent
43afe4696a
commit
d44546e8c6
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@ -683,6 +683,12 @@ dependencies = [
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"cfg-if",
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]
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[[package]]
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name = "ipnet"
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version = "2.4.0"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "35e70ee094dc02fd9c13fdad4940090f22dbd6ac7c9e7094a46cf0232a50bc7c"
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[[package]]
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name = "itertools"
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version = "0.10.3"
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@ -762,6 +768,7 @@ dependencies = [
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"rand",
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"rcgen",
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"ring",
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"rtcp",
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"rustls",
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"rustls-native-certs",
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"serde",
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@ -1063,6 +1070,17 @@ dependencies = [
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"paste",
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]
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[[package]]
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name = "parking_lot"
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version = "0.11.2"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "7d17b78036a60663b797adeaee46f5c9dfebb86948d1255007a1d6be0271ff99"
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dependencies = [
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"instant",
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"lock_api",
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"parking_lot_core 0.8.5",
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]
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[[package]]
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name = "parking_lot"
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version = "0.12.0"
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@ -1070,7 +1088,21 @@ source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "87f5ec2493a61ac0506c0f4199f99070cbe83857b0337006a30f3e6719b8ef58"
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dependencies = [
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"lock_api",
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"parking_lot_core",
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"parking_lot_core 0.9.1",
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]
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[[package]]
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name = "parking_lot_core"
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version = "0.8.5"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "d76e8e1493bcac0d2766c42737f34458f1c8c50c0d23bcb24ea953affb273216"
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dependencies = [
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"cfg-if",
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"instant",
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"libc",
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"redox_syscall",
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"smallvec",
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"winapi",
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]
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[[package]]
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@ -1272,6 +1304,17 @@ dependencies = [
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"winapi",
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]
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[[package]]
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name = "rtcp"
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version = "0.6.5"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "80d9625e47edb43aca711ec826ad12154d364ada9e60f4e6f8d40471b3e1e156"
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dependencies = [
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"bytes",
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"thiserror",
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"webrtc-util",
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]
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[[package]]
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name = "rustls"
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version = "0.20.4"
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@ -1653,6 +1696,7 @@ dependencies = [
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"mio",
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"num_cpus",
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"once_cell",
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"parking_lot 0.12.0",
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"pin-project-lite",
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"signal-hook-registry",
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"socket2",
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@ -1781,7 +1825,7 @@ version = "0.3.9"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "9e0ab7bdc962035a87fba73f3acca9b8a8d0034c2e6f60b84aeaaddddc155dce"
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dependencies = [
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"parking_lot",
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"parking_lot 0.12.0",
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"sharded-slab",
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"smallvec",
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"thread_local",
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@ -1996,6 +2040,28 @@ dependencies = [
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"webpki",
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]
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[[package]]
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name = "webrtc-util"
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version = "0.5.3"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "60b69bdea720881eddee50bd969be052ed95f04ccc5a6f684495131afb87e31c"
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dependencies = [
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"async-trait",
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"bitflags",
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"bytes",
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"cc",
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"ipnet",
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"lazy_static",
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"libc",
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"log",
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"nix",
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"parking_lot 0.11.2",
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"rand",
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"thiserror",
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"tokio",
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"winapi",
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]
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[[package]]
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name = "winapi"
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version = "0.3.9"
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@ -23,6 +23,7 @@ cocoa = { version = "0.24", default-features = false }
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[features]
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default = ["tls-rustls-native-roots"]
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log-rtp = ["lib-gst-meet/log-rtp"]
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tls-insecure = ["lib-gst-meet/tls-insecure"]
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tls-native = ["lib-gst-meet/tls-native"]
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tls-native-vendored = ["lib-gst-meet/tls-native-vendored"]
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@ -58,6 +58,12 @@ struct Opt {
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help = "Disable TLS certificate verification (use with extreme caution)"
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)]
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tls_insecure: bool,
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#[cfg(feature = "log-rtp")]
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#[structopt(
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long,
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help = "Log all RTP/RTCP packets at DEBUG level"
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)]
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log_rtp: bool,
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}
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#[cfg(not(target_os = "macos"))]
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@ -154,6 +160,8 @@ async fn main_inner() -> Result<()> {
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region,
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video_codec,
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recv_pipeline_participant_template,
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#[cfg(feature = "log-rtp")]
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log_rtp,
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..
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} = opt;
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@ -164,6 +172,10 @@ async fn main_inner() -> Result<()> {
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region,
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video_codec,
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extra_muc_features: vec![],
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start_bitrate: 800,
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stereo: false,
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#[cfg(feature = "log-rtp")]
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log_rtp,
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};
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let main_loop = glib::MainLoop::new(None, false);
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@ -17,3 +17,7 @@ tracing = { version = "0.1", default-features = false }
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[lib]
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name = "gstmeet"
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crate_type = ["staticlib", "cdylib"]
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[features]
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default = []
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log-rtp = ["lib-gst-meet/log-rtp"]
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@ -156,6 +156,13 @@ pub unsafe extern "C" fn gstmeet_connection_join_conference(
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.to_string_lossy()
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.to_string(),
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extra_muc_features: vec![],
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// TODO
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start_bitrate: 800,
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stereo: false,
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#[cfg(feature = "log-rtp")]
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log_rtp: false,
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};
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(*context)
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.runtime
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@ -32,6 +32,7 @@ pem = { version = "1", default-features = false }
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rand = { version = "0.8", default-features = false, features = ["std", "std_rng"] }
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rcgen = { version = "0.9", default-features = false }
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ring = { version = "0.16", default-features = false }
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rtcp = { version = "0.6", default-features = false, optional = true }
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rustls = { version = "0.20", default-features = false, features = ["logging", "tls12"], optional = true }
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rustls-native-certs = { version = "0.6", default-features = false, optional = true }
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serde = { version = "1", default-features = false, features = ["derive"] }
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@ -55,6 +56,7 @@ xmpp-parsers = { git = "https://gitlab.com/xmpp-rs/xmpp-rs.git", default-feature
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# Ideally we would enable rustls/dangerous_configuration only when tls-insecure is enabled, but until weak-dep-features is stabilised, that
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# would cause rustls to always be pulled in.
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default = ["tls-rustls-native-roots"]
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log-rtp = ["rtcp"]
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tls-insecure = []
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tls-native = ["tokio-tungstenite/native-tls", "native-tls"]
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tls-native-vendored = ["tokio-tungstenite/native-tls-vendored", "native-tls/vendored"]
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@ -75,6 +75,10 @@ pub struct JitsiConferenceConfig {
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pub region: Option<String>,
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pub video_codec: String,
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pub extra_muc_features: Vec<String>,
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pub start_bitrate: u32,
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pub stereo: bool,
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#[cfg(feature = "log-rtp")]
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pub log_rtp: bool,
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}
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#[derive(Clone)]
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@ -140,10 +144,8 @@ impl JitsiConference {
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machine_uid: Uuid::new_v4().to_string(),
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room: config.muc.to_string(),
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properties: hashmap! {
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// Disable voice processing
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// TODO put this in config
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"stereo".to_string() => "true".to_string(),
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"startBitrate".to_string() => "800".to_string(),
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"stereo".to_string() => config.stereo.to_string(),
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"startBitrate".to_string() => config.start_bitrate.to_string(),
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},
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};
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@ -5,6 +5,8 @@ use futures::stream::StreamExt;
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use glib::{ObjectExt, ToValue};
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use gstreamer::prelude::{ElementExt, GObjectExtManualGst, GstBinExt, PadExt};
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use gstreamer_rtp::{prelude::RTPHeaderExtensionExt, RTPHeaderExtension};
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#[cfg(feature = "log-rtp")]
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use gstreamer_rtp::RTPBuffer;
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use jitsi_xmpp_parsers::{
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jingle::{Action, Content, Description, Jingle, Transport},
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jingle_dtls_srtp::Fingerprint,
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@ -126,7 +128,6 @@ impl Codec {
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struct ParsedRtpDescription {
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codecs: Vec<Codec>,
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audio_hdrext_ssrc_audio_level: Option<u16>,
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audio_hdrext_transport_cc: Option<u16>,
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video_hdrext_transport_cc: Option<u16>,
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}
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@ -190,7 +191,6 @@ impl JingleSession {
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let mut vp8 = None;
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let mut vp9 = None;
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let mut audio_hdrext_ssrc_audio_level = None;
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let mut audio_hdrext_transport_cc = None;
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let mut video_hdrext_transport_cc = None;
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if description.media == "audio" {
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@ -209,9 +209,6 @@ impl JingleSession {
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if hdrext.uri == RTP_HDREXT_SSRC_AUDIO_LEVEL {
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audio_hdrext_ssrc_audio_level = Some(hdrext.id);
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}
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else if hdrext.uri == RTP_HDREXT_TRANSPORT_CC {
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audio_hdrext_transport_cc = Some(hdrext.id);
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}
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}
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}
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else if description.media == "video" {
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@ -274,8 +271,6 @@ impl JingleSession {
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}
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}
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for hdrext in description.hdrexts.iter() {
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// TODO: .parse::<u8>() won’t be needed after updating xmpp-parsers, it is now a u16 as
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// defined in the XEP and related RFC.
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if hdrext.uri == RTP_HDREXT_TRANSPORT_CC {
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video_hdrext_transport_cc = Some(hdrext.id);
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}
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@ -325,7 +320,6 @@ impl JingleSession {
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Ok(Some(ParsedRtpDescription {
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codecs,
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audio_hdrext_ssrc_audio_level,
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audio_hdrext_transport_cc,
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video_hdrext_transport_cc,
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}))
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}
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@ -467,7 +461,6 @@ impl JingleSession {
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let mut ice_transport = None;
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let mut codecs = vec![];
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let mut audio_hdrext_ssrc_audio_level = None;
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let mut audio_hdrext_transport_cc = None;
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let mut video_hdrext_transport_cc = None;
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let mut remote_ssrc_map = HashMap::new();
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@ -480,8 +473,6 @@ impl JingleSession {
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codecs.extend(description.codecs);
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audio_hdrext_ssrc_audio_level =
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audio_hdrext_ssrc_audio_level.or(description.audio_hdrext_ssrc_audio_level);
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audio_hdrext_transport_cc =
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audio_hdrext_transport_cc.or(description.audio_hdrext_transport_cc);
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video_hdrext_transport_cc =
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video_hdrext_transport_cc.or(description.video_hdrext_transport_cc);
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}
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@ -592,9 +583,6 @@ impl JingleSession {
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if let Some(hdrext) = audio_hdrext_ssrc_audio_level {
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caps = caps.field(&format!("extmap-{}", hdrext), RTP_HDREXT_SSRC_AUDIO_LEVEL);
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}
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if let Some(hdrext) = audio_hdrext_transport_cc {
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caps = caps.field(&format!("extmap-{}", hdrext), &RTP_HDREXT_TRANSPORT_CC);
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}
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}
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else {
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// A video codec, as the only audio codec we support is Opus.
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@ -829,14 +817,6 @@ impl JingleSession {
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let hdrext = RTPHeaderExtension::create_from_uri(&ext_uri)
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.context("failed to create hdrext")?;
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hdrext.set_id(ext_id);
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// if ext_uri == RTP_HDREXT_SSRC_AUDIO_LEVEL {
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// }
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// else if ext_uri == RTP_HDREXT_TRANSPORT_CC {
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// // hdrext.set_property("n-streams", 2u32);
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// }
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// else {
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// bail!("unknown rtp hdrext: {}", ext_uri);
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// };
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Ok::<_, anyhow::Error>(hdrext)
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};
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match f() {
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@ -941,14 +921,6 @@ impl JingleSession {
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let hdrext =
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RTPHeaderExtension::create_from_uri(&ext_uri).context("failed to create hdrext")?;
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hdrext.set_id(ext_id);
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// if ext_uri == RTP_HDREXT_SSRC_AUDIO_LEVEL {
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// }
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// else if ext_uri == RTP_HDREXT_TRANSPORT_CC {
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// // hdrext.set_property("n-streams", 2u32);
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// }
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// else {
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// bail!("unknown rtp hdrext: {}", ext_uri);
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// }
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Ok::<_, anyhow::Error>(hdrext)
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};
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match f() {
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@ -995,12 +967,6 @@ impl JingleSession {
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let hdrext =
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RTPHeaderExtension::create_from_uri(&ext_uri).context("failed to create hdrext")?;
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hdrext.set_id(ext_id);
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if ext_uri == RTP_HDREXT_TRANSPORT_CC {
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// hdrext.set_property("n-streams", 2u32);
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}
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else {
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bail!("unknown rtp hdrext: {}", ext_uri);
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}
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Ok::<_, anyhow::Error>(hdrext)
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};
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match f() {
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@ -1029,13 +995,79 @@ impl JingleSession {
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debug!("linking rtpfunnel -> rtpbin");
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rtpfunnel.link_pads(None, &rtpbin, Some("send_rtp_sink_0"))?;
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let rtp_recv_identity = gstreamer::ElementFactory::make("identity", None)?;
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pipeline.add(&rtp_recv_identity)?;
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let rtcp_recv_identity = gstreamer::ElementFactory::make("identity", None)?;
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pipeline.add(&rtcp_recv_identity)?;
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let rtp_send_identity = gstreamer::ElementFactory::make("identity", None)?;
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pipeline.add(&rtp_send_identity)?;
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let rtcp_send_identity = gstreamer::ElementFactory::make("identity", None)?;
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pipeline.add(&rtcp_send_identity)?;
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#[cfg(feature = "log-rtp")]
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if conference.config.log_rtp {
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let make_rtp_logger = |direction: &'static str| {
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move |values: &[glib::Value]| -> Option<glib::Value> {
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let f = || {
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let buffer: gstreamer::Buffer = values[1].get()?;
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let rtp_buffer = RTPBuffer::from_buffer_readable(&buffer)?;
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debug!(
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ssrc=rtp_buffer.ssrc(),
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seq=rtp_buffer.seq(),
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ts=rtp_buffer.timestamp(),
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marker=rtp_buffer.is_marker(),
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payload_size=rtp_buffer.payload_size(),
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"RTP {}",
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direction,
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);
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Ok::<_, anyhow::Error>(())
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};
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if let Err(e) = f() {
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warn!("RTP {}: {:?}", direction, e);
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}
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None
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}
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};
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let make_rtcp_logger = |direction: &'static str| {
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move |values: &[glib::Value]| -> Option<glib::Value> {
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let f = || {
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let buffer: gstreamer::Buffer = values[1].get()?;
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let mut buf = [0u8; 1500];
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buffer.copy_to_slice(0, &mut buf[..buffer.size()]).map_err(|_| anyhow!("invalid RTCP packet size"))?;
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let decoded = rtcp::packet::unmarshal(&mut &buf[..buffer.size()])?;
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debug!(
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"RTCP {} size={}\n{:?}",
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direction,
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buffer.size(),
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decoded,
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);
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Ok::<_, anyhow::Error>(())
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};
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if let Err(e) = f() {
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warn!("RTCP {}: {:?}", direction, e);
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}
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None
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}
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};
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rtp_recv_identity.connect("handoff", false, make_rtp_logger("RECV"));
|
||||
rtp_send_identity.connect("handoff", false, make_rtp_logger("SEND"));
|
||||
rtcp_recv_identity.connect("handoff", false, make_rtcp_logger("RECV"));
|
||||
rtcp_send_identity.connect("handoff", false, make_rtcp_logger("SEND"));
|
||||
}
|
||||
|
||||
debug!("link dtlssrtpdec -> rtpbin");
|
||||
dtlssrtpdec.link_pads(Some("rtp_src"), &rtpbin, Some("recv_rtp_sink_0"))?;
|
||||
dtlssrtpdec.link_pads(Some("rtcp_src"), &rtpbin, Some("recv_rtcp_sink_0"))?;
|
||||
dtlssrtpdec.link_pads(Some("rtp_src"), &rtp_recv_identity, None)?;
|
||||
rtp_recv_identity.link_pads(None, &rtpbin, Some("recv_rtp_sink_0"))?;
|
||||
dtlssrtpdec.link_pads(Some("rtcp_src"), &rtcp_recv_identity, None)?;
|
||||
rtcp_recv_identity.link_pads(None, &rtpbin, Some("recv_rtcp_sink_0"))?;
|
||||
|
||||
debug!("linking rtpbin -> dtlssrtpenc");
|
||||
rtpbin.link_pads(Some("send_rtp_src_0"), &dtlssrtpenc, Some("rtp_sink_0"))?;
|
||||
rtpbin.link_pads(Some("send_rtcp_src_0"), &dtlssrtpenc, Some("rtcp_sink_0"))?;
|
||||
rtpbin.link_pads(Some("send_rtp_src_0"), &rtp_send_identity, None)?;
|
||||
rtp_send_identity.link_pads(None, &dtlssrtpenc, Some("rtp_sink_0"))?;
|
||||
rtpbin.link_pads(Some("send_rtcp_src_0"), &rtcp_send_identity, None)?;
|
||||
rtcp_send_identity.link_pads(None, &dtlssrtpenc, Some("rtcp_sink_0"))?;
|
||||
|
||||
debug!("linking ice src -> dtlssrtpdec");
|
||||
nicesrc.link(&dtlssrtpdec)?;
|
||||
|
|
@ -1183,11 +1215,6 @@ impl JingleSession {
|
|||
RTP_HDREXT_SSRC_AUDIO_LEVEL.to_owned(),
|
||||
));
|
||||
}
|
||||
if let Some(hdrext) = audio_hdrext_transport_cc {
|
||||
description
|
||||
.hdrexts
|
||||
.push(RtpHdrext::new(hdrext, RTP_HDREXT_TRANSPORT_CC.to_owned()));
|
||||
}
|
||||
}
|
||||
else if initiate_content.name.0 == "video" {
|
||||
if let Some(hdrext) = video_hdrext_transport_cc {
|
||||
|
|
|
|||
Loading…
Reference in New Issue