Fix OpenSIPS + rtpengine RTP NAT Audio Issue (No Audio When SDP Has Private IP)
Budget: $30 – $250 USD
We are looking for an experienced VoIP engineer (OpenSIPS + rtpengine) to debug and fix a NAT-related audio issue in our production SIP platform.
We have OpenSIPS 3.3 + rtpengine 1.11 behind NAT.
Calls work when SDP contains public IP.
No audio when SDP contains private IP.
RTP visible on firewall but not on server.
Need an expert to debug NAT, firewall, and rtpengine behavior.
Must have real OpenSIPS + rtpengine experience.
Important:
This is NOT a basic SIP config task.
We need a senior VoIP engineer who understands deep NAT + RTP edge cases.
If you don’t have hands-on OpenSIPS + rtpengine experience, please do not apply.
We have OpenSIPS 3.3 + rtpengine 1.11 behind NAT.
Calls work when SDP contains public IP.
No audio when SDP contains private IP.
RTP visible on firewall but not on server.
Need an expert to debug NAT, firewall, and rtpengine behavior.
Must have real OpenSIPS + rtpengine experience.
Important:
This is NOT a basic SIP config task.
We need a senior VoIP engineer who understands deep NAT + RTP edge cases.
If you don’t have hands-on OpenSIPS + rtpengine experience, please do not apply.
Related categories:
Asterisk PBX
VoIP
Telecommunications Engineering
Software Development
FreeSwitch
Telecoms Engineering
Telecom
SIP