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support turn relay config, and force_relay option

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This commit is contained in:
lc
2025-11-16 04:04:25 +08:00
parent 483cf9d225
commit 0da5d379fc
3 changed files with 113 additions and 13 deletions
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2
examples/webrtc.rs
View File

@@ -70,7 +70,7 @@ async fn main() -> Result<()> {
"".to_string()
};
let webrtc_stream = WebRTCStream::new(&remote_endpoint, 30000).await?;
let webrtc_stream = WebRTCStream::new(&remote_endpoint, false, 30000).await?;
// Print the offer to be sent to the other peer
let local_endpoint = webrtc_stream.get_local_endpoint().await?;

2
src/socket_client.rs
View File

@@ -134,7 +134,7 @@ pub async fn connect_tcp<
#[cfg(feature = "webrtc")]
if is_webrtc_endpoint(&target.to_string()) {
return Ok(Stream::WebRTC(
webrtc::WebRTCStream::new(&target.to_string(), ms_timeout).await?,
webrtc::WebRTCStream::new(&target.to_string(), false, ms_timeout).await?,
));
}
let target_str = check_ws(&target.to_string());

122
src/webrtc.rs
View File

@@ -11,6 +11,7 @@ use webrtc::ice::mdns::MulticastDnsMode;
use webrtc::ice_transport::ice_server::RTCIceServer;
use webrtc::peer_connection::configuration::RTCConfiguration;
use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
use webrtc::peer_connection::policy::ice_transport_policy::RTCIceTransportPolicy;
use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
use webrtc::peer_connection::RTCPeerConnection;
@@ -20,7 +21,9 @@ use bytes::{Bytes, BytesMut};
use tokio::sync::watch;
use tokio::sync::Mutex;
use tokio::time::timeout;
use url::Url;
use crate::config;
use crate::protobuf::Message;
use crate::sodiumoxide::crypto::secretbox::Key;
use crate::ResultType;
@@ -36,7 +39,15 @@ pub struct WebRTCStream {
/// Most browsers, including Chromium, enforce this protocol limit.
const DATA_CHANNEL_BUFFER_SIZE: u16 = u16::MAX;
const DEFAULT_ICE_SERVER: &str = "stun:stun.cloudflare.com:3478";
// use 3 public STUN servers to find out the NAT type, 2 must be the same address but different ports
// https://stackoverflow.com/questions/72805316/determine-nat-mapping-behaviour-using-two-stun-servers
// luckily nextcloud supports two ports for STUN
// unluckily webrtc-rs does not use the same port to do the STUN request
static DEFAULT_ICE_SERVERS: [&str; 3] = [
"stun:stun.cloudflare.com:3478",
"stun:stun.nextcloud.com:3478",
"stun:stun.nextcloud.com:443",
];
lazy_static::lazy_static! {
static ref SESSIONS: Arc::<Mutex<HashMap<String, WebRTCStream>>> = Default::default();
@@ -126,7 +137,35 @@ impl WebRTCStream {
Self::get_key_for_sdp(&desc)
}
pub async fn new(remote_endpoint: &str, ms_timeout: u64) -> ResultType<Self> {
#[inline]
fn get_turn_server_from_url(url: &str) -> Option<RTCIceServer> {
// standard url format with turn scheme: turn://user:pass@host:port
match Url::parse(url) {
Ok(u) => {
if u.scheme() == "turn" {
Some(RTCIceServer {
urls: vec![format!(
"turn:{}:{}",
u.host_str().unwrap_or_default(),
u.port().unwrap_or(3478)
)],
username: u.username().to_string(),
credential: u.password().unwrap_or_default().to_string(),
..Default::default()
})
} else {
None
}
}
Err(_) => None,
}
}
pub async fn new(
remote_endpoint: &str,
force_relay: bool,
ms_timeout: u64,
) -> ResultType<Self> {
log::debug!("New webrtc stream to endpoint: {}", remote_endpoint);
let remote_offer = if remote_endpoint.is_empty() {
"".into()
@@ -144,6 +183,7 @@ impl WebRTCStream {
}
drop(sessions_lock);
let start_local_offer = remote_offer.is_empty();
// Create a SettingEngine and enable Detach
let mut s = SettingEngine::default();
s.detach_data_channels();
@@ -151,17 +191,29 @@ impl WebRTCStream {
// Create the API object
let api = APIBuilder::new().with_setting_engine(s).build();
let mut ice_servers = vec![RTCIceServer {
urls: DEFAULT_ICE_SERVERS.iter().map(|s| s.to_string()).collect(),
..Default::default()
}];
if start_local_offer {
// only offer needs TURN server
let relay_server = config::Config::get_option(config::keys::OPTION_RELAY_SERVER);
if let Some(turn_server) = Self::get_turn_server_from_url(&relay_server) {
ice_servers.push(turn_server);
}
}
// Prepare the configuration
let config = RTCConfiguration {
ice_servers: vec![RTCIceServer {
urls: vec![DEFAULT_ICE_SERVER.to_string()],
..Default::default()
}],
ice_servers,
ice_transport_policy: if force_relay {
RTCIceTransportPolicy::Relay
} else {
RTCIceTransportPolicy::All
},
..Default::default()
};
let start_local_offer = remote_offer.is_empty();
let (notify_tx, notify_rx) = watch::channel(false);
// Create a new RTCPeerConnection
let pc = Arc::new(api.new_peer_connection(config).await?);
@@ -409,6 +461,54 @@ mod tests {
use crate::webrtc::WebRTCStream;
use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
#[test]
fn test_webrtc_turn_url() {
assert_eq!(
WebRTCStream::get_turn_server_from_url("turn://example.com:3478")
.unwrap_or_default()
.urls[0],
"turn:example.com:3478"
);
assert_eq!(
WebRTCStream::get_turn_server_from_url("turn://example.com")
.unwrap_or_default()
.urls[0],
"turn:example.com:3478"
);
assert_eq!(
WebRTCStream::get_turn_server_from_url("turn://123@example.com")
.unwrap_or_default()
.username,
"123"
);
assert_eq!(
WebRTCStream::get_turn_server_from_url("turn://123@example.com")
.unwrap_or_default()
.credential,
""
);
assert_eq!(
WebRTCStream::get_turn_server_from_url("turn://123:321@example.com")
.unwrap_or_default()
.credential,
"321"
);
assert_eq!(
WebRTCStream::get_turn_server_from_url("stun://example.com:3478"),
None
);
assert_eq!(
WebRTCStream::get_turn_server_from_url("http://123:123@example.com:3478"),
None
);
}
#[test]
fn test_webrtc_session_key() {
let mut sdp_str = "".to_owned();
@@ -554,18 +654,18 @@ IHR5cCBzcmZseCByYWRkciAwLjAuMC4wIHJwb3J0IDY0MDA4XHJcbmE9ZW5kLW9mLWNhbmRpZGF0ZXNc
async fn test_webrtc_new_stream() {
let mut endpoint = "webrtc://sdfsdf".to_owned();
assert!(
WebRTCStream::new(&endpoint, 10000).await.is_err(),
WebRTCStream::new(&endpoint, false, 10000).await.is_err(),
"invalid webrtc endpoint should error"
);
endpoint = "wss://sdfsdf".to_owned();
assert!(
WebRTCStream::new(&endpoint, 10000).await.is_err(),
WebRTCStream::new(&endpoint, false, 10000).await.is_err(),
"invalid webrtc endpoint should error"
);
assert!(
WebRTCStream::new("", 10000).await.is_ok(),
WebRTCStream::new("", false, 10000).await.is_ok(),
"local webrtc endpoint should ok"
);
@@ -580,7 +680,7 @@ LjIgNjQwMDcgdHlwIGhvc3RcclxuYT1jYW5kaWRhdGU6MTg2MTA0NTE5MCAxIHVkcCAxNjk0NDk4ODE1
ggcmFkZHIgMC4wLjAuMCBycG9ydCA2NDAwOFxyXG5hPWNhbmRpZGF0ZToxODYxMDQ1MTkwIDIgdWRwIDE2OTQ0OTg4MTUgMTQuMjEyLjY4LjEyIDI3MDA0\
IHR5cCBzcmZseCByYWRkciAwLjAuMC4wIHJwb3J0IDY0MDA4XHJcbmE9ZW5kLW9mLWNhbmRpZGF0ZXNcclxuIn0=".to_owned();
assert!(
WebRTCStream::new(&endpoint, 10000).await.is_err(),
WebRTCStream::new(&endpoint, false, 10000).await.is_err(),
"connect to an 'answer' webrtc endpoint should error"
);
}