No Audio on Outbound SIP Call (Go SDK, Windows, Plivo) - Frames Sent but Silence Received Hi everyone, I'm building a voice agent using the LiveKit Go SDK and facing a persistent "no audio" issue on outbound SIP calls (Plivo). I'd appreciate any insights! Setup: • OS: Windows 11 (Running natively, no WSL2). • SDK:

<http://github.com/livekit/server-sdk-go/v2|github.com/livekit/server-sdk-go/v2>

• SIP Provider: Plivo (Outbound Trunk). • Flow: Worker connects to room -> Waits for SIP participant -> Publishes Audio Track -> Sends TTS audio. The Issue: The call connects successfully, and the SIP participant (phone) answers. The worker receives audio from the phone perfectly. The worker also sends audio (logs show frames being encoded and sent), but the phone hears absolute silence. What I've Tried: 1. Opus Codec: Standard

webrtc.MimeTypeOpus

(48kHz). Result: Silence. 2. G.711 PCMA: Switched to manual G.711 A-law encoding (8kHz, 20ms frames) using

webrtc.MimeTypePCMA

. Result: Silence. 3. Network: Forced

ICETransportPolicyRelay

to rule out NAT/Firewall issues. Connection is stable (

ICE connected

). Logs (Worker Side): It looks like the track is published and data is flowing:

"message":"[AUDIO] publishing PCMA audio track (G.711 A-law)"
"message":"published track", "name"="agent-audio", "source"="MICROPHONE"
"message":"[AUDIO] PCMA frame encoded", "pcma_bytes":160, "payload_preview":"d52aaad5..."
"message":"[AUDIO] pacing progress", "frames_sent":100
"message":"[AUDIO] pacing progress", "frames_sent":200

Code Snippet (Audio Publishing): I'm using

TrackLocalStaticSample

and writing 20ms frames manually.

// Publishing the track
track, err := webrtc.NewTrackLocalStaticSample(
    webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypePCMA},
    "agent-audio",
    "agent-stream",
)
_, err = room.LocalParticipant.PublishTrack(track, &lksdk.TrackPublicationOptions{
    Name: "agent-audio",
})

// Writing Audio (Pacing Loop)
// Sending 160-byte frames every 20ms (8kHz A-law)
sample := media.Sample{
    Data:     audioFrame,
    Duration: 20 * time.Millisecond,
}
err := track.WriteSample(sample)

SIP Config (Plivo):

{
  "provider": "PLIVO",
  "transport": "tcp",
  "codecs": ["PCMA", "PCMU", "G722"] // (Assumed defaults)
}

Questions: 1. Are there known issues with

TrackLocalStaticSample

for SIP outbound? 2. Does the SIP Gateway require specific RTP header extensions or StreamIDs to recognize the audio? 3. Could this be an encryption mismatch (SRTP) even if the connection is established? Any help debugging this would be amazing. Thanks!

Not sure I am following exactly. Is this a self hosted LiveKit server or are you using LiveKit cloud? If using LiveKit cloud can you share a call ID that had the issue so I can take a look at the logs and PCAP.

its livekit cloud

I think something is wrong with the opus encoding

Error: *opus: buffer too small*

Thnx

@refined-appointment-81829 can you share the logs of this call id SCL_X6TuFTCZrUR4

You should be able to find the call and PCAP on your dashboard: https://cloud.livekit.io/projects/p_/telephony

Ok