About Education
The SIP training, Session Initiation Protocol (SIP) training course offered by TONEX provides an overview of SIP, its components and how it works. It covers data networking principles to telco engineers and signalling principles to IP engineers.
Prerequisites
Working knowledge of converged voice, video and data networks
Duration of Training
- Training with instructor: 3 days, with practical laboratory work
- Virtual instructor-led training: 3 days, with web-based lectures and hands-on laboratory work
Who Should Participate?
- VoIP Engineers
- Network engineers
- System engineers
- Telco operatörleri
- IT Experts
- Cybersecurity experts
- University students (communications, computer science)
Educational Content
- Chapter 1: SIP Overview
- The evolution of VoIP technology
- Comparison with traditional telephone systems
- Advantages and disadvantages of SIP
- Chapter 2: SIP Components
- User Agent (UA)
- User Agent Server (UAS)
- Proxy Server
- Redirector Server
- Registration Server
- Location Server
- Chapter 3: Basic Call Flow
- Registration process
- Call setup
- Call continued
- Call termination
- Chapter 4: Using Wireshark for SIP Analysis
- Capturing and analyzing SIP packets with Wireshark
- Examining different fields of SIP messages
- Chapter 5: RTP Analysis
- Transporting audio/video packets with RTP protocol
- Analysis of RTP packets
- Chapter 6: SIP Methods
- Explanation of methods such as INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, INFO, SUBSCRIBE, NOTIFY
- Chapter 7: SDP Overview
- Carrying media information such as audio/video codecs, network information, timing information
- Structure of SDP messages
- Chapter 8: SIP Headers
- Function of headers such as To, From, Via, Call-ID, CSeq, Contact, Content-Type, Content-Length
- Chapter 9: Response Codes
- Meanings of response codes in the series 1xx, 2xx, 3xx, 4xx, 5xx
- Chapter 10: DTMF
- Transmission and processing of DTMF tones
- Chapter 11: SIP Nat/Stun/Turn/Ice
- NAT traversal problems and solutions
- Function of STUN, TURN and ICE protocols
- Chapter 12: SIP Forking
- Sending requests to multiple servers simultaneously
- Advantages and disadvantages of forking
- Chapter 13: SIP Security
- SRTP (Secure Real-time Transport Protocol)
- TLS/SSL encryption
- Authentication mechanisms
- Chapter 14: WebRTC
- Key features of WebRTC technology
- Relationship between SIP and WebRTC
- Chapter 15: DNS/ENUM
- The role of DNS and ENUM records in SIP
- Username to IP address translation
- Chapter 16: ITSP
- Internet Telephone Service Providers (ITSP)
- Features of ITSP services
- Chapter 17: SIP Trunk and SBC
- SIP Trunk technology and its advantages
- Session Border Controller (SBC) devices and their functions
- Chapter 18: SIP Attacks
- Denial-of-Service (DoS) attacks
- Spoofing attacks
- Eavesdropping attacks
- Preventive measures
What You Will Gain at the End of Training
After completing this course you should be able to:
- Understanding VoIP basics
- Discover where, why and how SIP is used
- Understanding SIP basics
- Understanding SIP architecture and components