Question 1

Can OpenSIPS integrate WebRTC with existing PBX systems?

Accepted Answer

Yes, OpenSIPS serves as a bridge between WebRTC clients and traditional PBX systems by translating WebSocket signaling to standard SIP protocols. It can handle codec negotiation between WebRTC's limited codec support and PBX requirements, manage transport protocol differences, and coordinate authentication between different systems.

Question 2

How does OpenSIPS handle WebRTC security requirements?

Accepted Answer

OpenSIPS implements WebRTC security through mandatory WSS (secure WebSocket) connections, DTLS-SRTP media encryption, and origin validation.

Question 3

What is the difference between OpenSIPS and traditional WebRTC gateways?

Accepted Answer

OpenSIPS provides advanced SIP proxy functionality with WebRTC support, offering sophisticated routing logic, load balancing, and enterprise security features. Traditional WebRTC gateways typically provide basic protocol translation without the advanced call control and scalability features that OpenSIPS delivers for enterprise deployments.

Question 4

How to troubleshoot WebRTC call quality issues in OpenSIPS?

Accepted Answer

You can troubleshoot WebRTC call quality by analyzing the complete media path from the browser through OpenSIPS to the destination. Check codec negotiation logs, monitor RTP statistics for packet loss and jitter, verify STUN/TURN server connectivity, and correlate browser audio processing settings with network-level performance metrics.

Question 5

What are the challenges in OpenSIPS WebRTC integration?

Accepted Answer

The main challenges that you may face are NAT traversal (ICE, STUN/TURN), media negotiation (SRTP vs RTP), and codec mismatches. TLS setup and WebSocket timeouts are also common pitfalls.

OpenSIPS WebRTC Integration

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