I will configure, develop and troubleshoot your freeswitch or asterisk voip system

Facing SIP, VoIP, FreeSWITCH, Asterisk, RTP or WebRTC issues?

I can help you diagnose and fix them quickly.

I'm an experienced VoIP & Telecom Engineer specialising in SIP, RTP, FreeSWITCH, Asterisk, Kamailio, OpenSIPS, RTPengine and WebRTC.

I can troubleshoot:

• SIP registration & authentication issues
• One-way / no audio
• RTP and media flow problems
• Call drops & failed calls
• SIP 4xx/5xx errors
• NAT & firewall issues
• SIP trunk problems
• FreeSWITCH & Asterisk configuration
• WebRTC, WSS, STUN/TURN & ICE issues
• Codec & DTMF problems
• Call routing & dialplan issues
• Kamailio/OpenSIPS routing
• RTPengine configuration
• Docker/AWS VoIP issues
• SIP trace and log analysis

Technologies:

FreeSWITCH | Asterisk | Kamailio | OpenSIPS | RTPengine | SIP | RTP | WebRTC | SIP.js | PJSIP | Docker | AWS

I analyse your SIP logs, configuration, signaling and media flow to identify the root cause rather than applying random fixes.

Whether your calls are failing, dropping, connecting without audio, or having NAT/RTP problems, I can help get your VoIP system working reliably.

Please contact me before ordering complex or production-critical troubleshooting so I can review your requirements.