Recently, I worked on a FusionPBX + FreeSWITCH call connectivity issue involving browser-based WebRTC extensions where one side of the call could hear audio while the other side could not.
These problems can be challenging because the SIP signaling may look perfectly healthy while the actual RTP/media path is failing.
🔍 My troubleshooting approach includes:
✅ SIP signaling & SDP analysis
✅ RTP media flow verification
✅ NAT / STUN / TURN configuration
✅ FreeSWITCH Sofia profiles
✅ WebRTC & DTLS-SRTP
✅ ICE candidate analysis
✅ AWS EC2 networking & security groups
✅ Chrome webrtc-internals diagnostics
✅ One-way / no-audio troubleshooting
✅ Browser ↔ SIP ↔ PSTN media-path analysis
The key is to identify where the audio actually stops rather than repeatedly changing random SIP/NAT settings.
If you're running FusionPBX, FreeSWITCH, Asterisk, WebRTC, SIP, CPaaS, or
Voice AI infrastructure and experiencing:
🔸 One-way audio
🔸 No audio
🔸 Calls dropping
🔸 NAT-related issues
🔸 WebRTC connectivity problems
🔸 Poor voice quality
🔸 SIP/RTP failures
🔸 AWS VoIP connectivity issues
I can help diagnose the problem at the SIP + RTP + WebRTC + infrastructure level and work toward a production-ready solution.
📩 Having a VoIP issue that you can't isolate? Send me the symptoms, architecture, or logs.
#FreeSWITCH #FusionPBX #Asterisk #VoIP #WebRTC #SIP #RTP #VoiceAI #CPaaS #Telephony #SIPTrunking #AWS #EC2 #CloudTelephony #VoIPConsulting #VoiceTechnology #ContactCenter #RealTimeCommunication
If you're exploring similar solutions or want to implement this in your business, I’m open to a quick discussion. Let’s evaluate your use case and identify the most efficient approach to get results.