In this course, you will learn core concepts of how the Internet Protocol (IP) carries a Voice over IP (VoIP) packet. You will learn the fundamentals of Session Initiation Protocol (SIP) architecture, SIP-related IP services, the advantages and disadvantages of SIP Trunking as well as Quality of Service (QoS)-Related Protocol.
In the Classroom Live version of this course, you will gain proficiency with some of the most popular VoIP software and hardware, such as Wireshark, Asterisk PBX, Kamailio SIP Proxy, Linksys Ethernet phone, and SIP-based ATA in a hands-on labs. You will also cover Cisco QoS policy administration and demonstrate successful VoIP calls in high data traffic conditions.
What You'll Learn
- Core concepts of how Internet Protocol (IP) carries a VoIP packet
- Advantages and disadvantages of SIP Trunking
- Configure DHCP and DNS to support IP telephony
- Real-Time Transport Protocol (RTP)
- Session Initiation Protocol (SIP) - Call set up, Instant Messaging, Presence
- Session Description Protocol (SDP)
- SIP proxy, Session Border Controller (SBC), and SIP softswitch
- Media Gateway Control Protocol (MGCP) analysis
- MGCP architecture
- How to implement QoS to ensure the highest voice quality over your IP networks
- The impact of jitter, latency, and packet loss on VoIP networks
- How to use Wireshark to decode and troubleshoot RTP, SIP, and MGCP call flows
- Configure the trixbox Softswitch and SIP proxy
- Configure SIP gateways and softphones.
Who Should Attend
This class is for people who need to understand VoIP technology. IT managers, technical sales/marketing personnel, consultants, network designers and engineers, product design engineers developing integrated-services products, telecom technicians and managers integrating PBX services within data networks, and systems administrators who will manage a converged network would benefit from this course.