Learn SIP and WebRTC for real-time VoIP communication. Deploy, secure, and troubleshoot enterprise-grade VoIP systems.
Understand the fundamentals of SIP, including its role in VoIP systems.
Configure and troubleshoot SIP registrars, proxies, and endpoints.
Analyze SIP call flows and packets using tools like Wireshark and Termshark.
Integrate WebRTC with SIP for real-time communications.
SIP and WebRTC: Comprehensive VoIP Training is a 5-day hands-on course designed to teach participants how to deploy, secure, and troubleshoot Voice over IP (VoIP) systems using SIP (Session Initiation Protocol) and WebRTC (Web Real-Time Communications). Combining focused lectures with practical labs, this course provides a deep dive into SIP messages, headers, call flows, and the integration of WebRTC for real-time communications. Students will work with tools like Wireshark and Termshark to analyze SIP packets, set up SIP registrars and proxies, and configure VoIP systems using platforms like Kamailio, RTPEngine, and Asterisk. Additional topics include NAT traversal, SIP security best practices, and media transport protocols like RTP and SDP. By the end of the course, participants will be equipped with the skills to implement and troubleshoot enterprise-grade VoIP systems.
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Aaron Steele
Casey Pense
Chris Tsantiris
Javier Martin
Justin Gilley
Kathy Le
Kelson Smith
Oussama Azzam
Pascal Rodmacq
Randall Granier
Aaron Steele
Casey Pense
Chris Tsantiris
Javier Martin
Justin Gilley
Kathy Le
Kelson Smith
Oussama Azzam
Pascal Rodmacq
Randall Granier