WebRTC-SIP Gateway for MagnusBilling
Budget: $250 – $750 USD
I’m looking for an experienced VoIP / WebRTC / OpenSIPS engineer to set up a WebRTC ↔ SIP gateway connected to my existing MagnusBilling system.
The objective is to allow users to make and receive calls directly from a web browser using their current SIP credentials, while MagnusBilling continues handling real-time billing, call tracking, and balance control.
Tasks:
Install and configure:
OpenSIPS (as WebRTC–SIP proxy, mid_registrar mode)
RTPengine (for media translation between WebRTC and RTP)
coturn (for STUN/TURN and NAT traversal)
Nginx + SSL (for secure WSS/HTTPS access)
Develop a simple web softphone (using JsSIP or similar) with the following features:
SIP login and registration
Make and receive calls in the browser
Mute and unmute functionality
Music on Hold (MOH) option
DTMF tone support during active calls
Display of call status and connection state
Ensure that all calls are properly routed through MagnusBilling for:
Real-time billing updates
Call detail records (CDR)
Correct balance management
Deliverables:
Fully working WebRTC ↔ SIP gateway
Web softphone interface with Mute / MOH / DTMF support
Documentation covering main configuration files and installation steps
Live test confirming correct integration with MagnusBilling
Requirements:
Strong experience with OpenSIPS and RTPengine
Solid understanding of WebRTC (DTLS-SRTP, ICE, WSS)
Experience with SIP, RTP, DTMF, and NAT
Familiarity with MagnusBilling or similar VoIP billing systems
Proficiency in Linux server administration and Nginx SSL configuration
Timeline: 2–5 days (including configuration and testing)
Budget: Negotiable, based on experience and project scope
Please apply only if you have prior experience building WebRTC-to-SIP solutions or integrating OpenSIPS with billing systems like MagnusBilling.
The objective is to allow users to make and receive calls directly from a web browser using their current SIP credentials, while MagnusBilling continues handling real-time billing, call tracking, and balance control.
Tasks:
Install and configure:
OpenSIPS (as WebRTC–SIP proxy, mid_registrar mode)
RTPengine (for media translation between WebRTC and RTP)
coturn (for STUN/TURN and NAT traversal)
Nginx + SSL (for secure WSS/HTTPS access)
Develop a simple web softphone (using JsSIP or similar) with the following features:
SIP login and registration
Make and receive calls in the browser
Mute and unmute functionality
Music on Hold (MOH) option
DTMF tone support during active calls
Display of call status and connection state
Ensure that all calls are properly routed through MagnusBilling for:
Real-time billing updates
Call detail records (CDR)
Correct balance management
Deliverables:
Fully working WebRTC ↔ SIP gateway
Web softphone interface with Mute / MOH / DTMF support
Documentation covering main configuration files and installation steps
Live test confirming correct integration with MagnusBilling
Requirements:
Strong experience with OpenSIPS and RTPengine
Solid understanding of WebRTC (DTLS-SRTP, ICE, WSS)
Experience with SIP, RTP, DTMF, and NAT
Familiarity with MagnusBilling or similar VoIP billing systems
Proficiency in Linux server administration and Nginx SSL configuration
Timeline: 2–5 days (including configuration and testing)
Budget: Negotiable, based on experience and project scope
Please apply only if you have prior experience building WebRTC-to-SIP solutions or integrating OpenSIPS with billing systems like MagnusBilling.