serve.zone/siprouter
3
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

fix(proxy-engine): use negotiated SDP payload types when wiring SIP legs and enable default nnnoiseless features for telephony denoising

Browse Source
This commit is contained in:
Jürgen Kunz
2026-04-10 17:14:14 +00:00
parent a02146633b
commit c63a759689
11 changed files with 254 additions and 13 deletions
Download Patch File Download Diff File Expand all files Collapse all files

22
rust/crates/proxy-engine/src/call_manager.rs
View File

@@ -196,7 +196,17 @@ impl CallManager {
};
// Mutable borrow on call/leg is now released.
let sip_pt = codecs.first().copied().unwrap_or(9);
let mut sip_pt = codecs.first().copied().unwrap_or(9);
// If the message has SDP (e.g., 200 OK answer), use the negotiated codec
// instead of the offered one.
if msg.has_sdp_body() {
if let Some(ep) = parse_sdp_endpoint(&msg.body) {
if let Some(pt) = ep.codec_pt {
sip_pt = pt;
}
}
}
match action {
SipLegAction::None => {}
@@ -301,12 +311,20 @@ impl CallManager {
);
// Wire device leg to mixer.
// Use the device's preferred codec from its INVITE SDP,
// not the provider's negotiated codec.
let dev_pt = device_invite
.has_sdp_body()
.then(|| parse_sdp_endpoint(&device_invite.body))
.flatten()
.and_then(|ep| ep.codec_pt)
.unwrap_or(sip_pt);
if let Some(dev_remote_addr) = dev_remote {
let dev_channels = create_leg_channels();
spawn_sip_inbound(dev_rtp_socket.clone(), dev_channels.inbound_tx);
spawn_sip_outbound(dev_rtp_socket, dev_remote_addr, dev_channels.outbound_rx);
if let Some(call) = self.calls.get(call_id) {
call.add_leg_to_mixer(&dev_leg_id, sip_pt, dev_channels.inbound_rx, dev_channels.outbound_tx)
call.add_leg_to_mixer(&dev_leg_id, dev_pt, dev_channels.inbound_rx, dev_channels.outbound_tx)
.await;
}
}

4
rust/crates/proxy-engine/src/mixer.rs
View File

@@ -368,8 +368,8 @@ async fn mixer_loop(
.unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
};
// Per-leg inbound denoising at 48kHz.
// Skip for Opus/WebRTC legs — browsers already apply
// their own noise suppression via getUserMedia.
// Only for SIP telephony legs — WebRTC browsers
// already apply noise suppression via getUserMedia.
let processed = if slot.codec_pt != codec_lib::PT_OPUS {
TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
} else {