Why Asterisk Voicebridge ExternalMedia Audio Becomes One-Way (Root Causes & Fixes)

Quick links

• VoiceBridge Architecture
• Send audio back using ExternalMedia (RTP guide)
• NAT/Firewall duplex issues
• RTP direction, SSRC, payload & timing

This article is written to be practical: you can run each test, and you can map the fix to a concrete setting or code location.

Symptom A

You can hear the caller, caller cannot hear you.

• Asterisk is successfully sending RTP to your app.
• Your app is not sending RTP back correctly to the port Asterisk is listening on, or the bridge is wrong.

Symptom B

Caller can hear you, you cannot hear the caller.

• Your app is successfully injecting RTP to Asterisk.
• Asterisk is not sending RTP to your app (wrong external_host, NAT/firewall, codec mismatch, or directmedia bypass).

Root cause #1: You are sending RTP back to the wrong port (or you never learned the port)

ExternalMedia is not “send RTP to external_host and you're done”. Asterisk chooses a local RTP port for that ExternalMedia channel. Your app must send RTP to that port.

In VoiceBridge code, this is handled by reading ARI channel variables: UNICASTRTP_LOCAL_ADDRESS and UNICASTRTP_LOCAL_PORT.

// mylinehub-voicebridge/src/main/java/.../ExternalMediaManagerImpl.java
// getVariable(channelId, "UNICASTRTP_LOCAL_ADDRESS")
// getVariable(channelId, "UNICASTRTP_LOCAL_PORT")

If you never query these variables (or you ignore them), you will very often send RTP to the wrong place.

Root cause #2: ExternalMedia channel is in the wrong bridge type

Some bridge types or call states produce behavior that looks like “media is connected” but only one direction flows. For duplex AI voice, you normally want a mixing bridge and you must add the correct channels to it.

VoiceBridge explicitly creates and uses mixing bridges and controls which channel is “listen-only” vs “talk-only” using two ExternalMedia channels + a snoop channel strategy.

Root cause #3: The caller leg is not actually in Stasis / your dialplan bypasses the bridge

ExternalMedia only helps if the caller channel is managed under ARI and placed in the correct bridge. If your dialplan lets the call continue in a normal dialplan flow (or direct media is enabled), the media path might bypass your ExternalMedia bridge.

Fix: ensure your dialplan sends the channel into your Stasis app (the name must match your ARI application), and verify with ARI events that the channel entered Stasis.

Root cause #4: NAT or firewall breaks one direction (most common in production)

RTP is UDP. Many “it works locally” projects fail as soon as there is: a cloud VM, a home router, a symmetric NAT, or strict firewall.

• VoiceBridge host cannot reach Asterisk RTP ports
• Asterisk cannot reach VoiceBridge UDP port range (inbound)
• External host IP is private/unroutable from Asterisk
• SIP ALG / NAT rewriting creates mismatched RTP targets

See: Common NAT & firewall issues that break duplex audio.

Root cause #5: Codec/payload mismatch (PCMU vs PCM16 vs Opus)

ExternalMedia's format must match what you actually send. If Asterisk expects ulaw/PCMU but your app injects PCM16 raw audio (or wrong RTP payload type), you get “silence” (which users perceive as one-way audio).

Deep reference: G.711 u-law vs PCM16 vs Opus.

Root cause #6: RTP timing and SSRC rules are violated

Asterisk is strict about RTP: sequence numbers, timestamps, packet pacing, and SSRC changes. Many “send RTP back” examples send packets too fast, too slow, or with broken timestamps.

Reference: RTP direction, SSRC, payload & timing.

Fix A: Use two ExternalMedia channels and control direction explicitly

In the VoiceBridge call setup, it creates: extIn (Asterisk → VoiceBridge RTP) and extOut (VoiceBridge → Asterisk RTP), then uses proper bridges so the caller always receives injected audio.

// mylinehub-voicebridge/src/main/java/com/mylinehub/voicebridge/ari/impl/AriBridgeImpl.java
// createBridge(..., "mixing")
// createExternalMediaChannel(..., extIn)
// createExternalMediaChannel(..., extOut)
// createSnoopChannel(...) for capture/control

This is why VoiceBridge can implement real barge-in and stable duplex under RTP drift.

Fix B: Always read the negotiated RTP peer (UNICASTRTP_*) before injecting audio

The most common one-way mistake is “sending RTP back to the wrong place”. VoiceBridge queries the ExternalMedia channel variables and builds the correct RTP target for injection.

// mylinehub-voicebridge/src/main/java/com/mylinehub/voicebridge/ari/impl/ExternalMediaManagerImpl.java
UnicastRtpPeer peer = getUnicastRtpPeer(channelId);
// peer.host = UNICASTRTP_LOCAL_ADDRESS
// peer.port = UNICASTRTP_LOCAL_PORT

This is the difference between a demo and a production bridge.

Fix C: DB-driven runtime config prevents “works on my machine” audio bugs

VoiceBridge does not depend on scattered JSON configs. Runtime values (stasis app config, RTP ports, hostnames, AI instruction blocks) are loaded from PostgreSQL, so your deployment can be corrected without rebuilding binaries.

See: VoiceBridge DB-driven configuration: stasis_app_config + instructions.

Fix D: ARI + firewall rules are treated as part of the product

One-way audio is usually not “ARI is broken”; it is mis-exposed ports: ARI HTTP/WebSocket, RTP ranges, and host routing. MYLINEHUB documents this and provides production-safe defaults.

Reference: Ports required for FreePBX + Asterisk.

Wireshark checklist

• Confirm two RTP streams exist (Asterisk→app and app→Asterisk)
• Payload type matches codec (PCMU/PCMA/etc.)
• Timestamps increment consistently (no huge jumps/drift)
• SSRC stays stable (or changes predictably)
• No ICMP “Port unreachable” bursts

Full workflow: Wireshark live monitoring for SIP & RTP.

Asterisk-side checks

asterisk -rvvvvv

# See active channels (confirm ExternalMedia channels exist)
core show channels concise

# Check RTP settings and strict RTP behavior
pjsip show settings

# Turn on RTP debugging briefly (careful: noisy)
rtp set debug on
rtp set debug off

If RTP debug shows only “Sent RTP packet…” but not “Got RTP packet…”, your app isn’t reaching Asterisk’s chosen port.

Key files referenced in this article

• ExternalMedia creation + UNICASTRTP peer lookup: ExternalMediaManagerImpl.java
• Duplex bridge strategy (two ExternalMedia + snoop + mixing bridge): AriBridgeImpl.java
• Spring Boot runtime basics: mylinehub-voicebridge/src/main/resources/application.properties

Next article to read

• How to Send Audio Back to Caller Using ARI ExternalMedia
• Snoop vs ExternalMedia vs AudioSocket (true duplex comparison)
• How VoiceBridge achieves full duplex in production