Question 1

What is Kamailio, and how does it benefit WebRTC and VoIP deployments?

Accepted Answer

Kamailio is an open-source SIP server that enhances WebRTC and VoIP deployments by providing secure, scalable signaling and media handling. It's ideal for managing large-scale communication systems, offering optimized performance for real-time interactions.

Question 2

How does Kamailio optimize WebRTC gateways?

Accepted Answer

Kamailio bridges WebRTC and VoIP networks, simplifying protocol conversion, session handling, and media traversal. It also ensures secure, seamless communication with features like NAT traversal and support for TLS and SRTP.

Question 3

Why is Kamailio ideal for large-scale VoIP deployments?

Accepted Answer

Kamailio efficiently handles high call volumes and concurrent sessions with load balancing, clustering, and resource management. It is ideal for enterprises looking to scale VoIP deployments without compromising performance.

Question 4

Can Kamailio improve scalability in VoIP and WebRTC systems?

Accepted Answer

Kamailio improves scalability by distributing traffic across servers, managing resources, and supporting custom routing. It ensures smooth operation even under high VoIP and WebRTC systems loads.

Question 5

What security features does Kamailio offer for VoIP and WebRTC?

Accepted Answer

Kamailio offers encryption with TLS, DTLS, and SRTP, along with AAA services for secure access. It also includes protection against SIP flooding and DoS attacks, making it a secure solution for real-time communication.

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Kamailio WebRTC Optimization and VoIP Deployments