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Duration 21 hours
Course Outline
Part I: Introduction
- Introduction
- Historical context and motivation
- Types of VoIP and their evolution
- SIP core concepts
- SIP standardization (RFC 3261 and other relevant standards)
- Architecture
- UA – User Agent
- Predefined servers: Registrar, Location, Proxy, and Redirect
- Application servers
- Identification and addressing mechanisms
- The SIP trapezoid model
- Servers and their operation
- Registration processes
- SIP server operations in Proxy and Redirect modes
- Stateless and stateful Proxy servers
- Location server functionality
- SRV records and DNS integration
- URI/URL/URN, ENUM, and NAPTR records
- SIP signalling messages (including Instant Messaging & Presence – IMP extensions)
- Message structure
- Request types
- Response types
- Example call sequences
- Headers and parameters
- IMP models
- SDP (Session Description Protocol)
- Media description
- Standard list of codecs
- Session negotiation rules
- Call flows – SIP signalling
- SIP session overview – primary RFC 3261 example
- Sample call scenarios
- Conferencing and IP PBX integration
- Modifying media during a session
- Utilizing IMP
- Routing of SIP requests and responses
- VIA header usage
- ROUTE and RECORD-ROUTE headers
- SIP-PSTN interworking
- SIP-T and SIP-I protocols
- SIP early media and SIP trunking
- SIP-PSTN signalling interactions
- SIP – security challenges
- Secure SIP, Secure RTP, and Secure RTCP
- Typical implementations of Secure SIP
- Practical problems and future perspectives
- NAT and firewall traversal
- Quality of Service (QoS)
- SIP and SDP within 3GPP IMS architecture
- Conclusion and discussion
Part II: Hands On
- SIP in LAN environment: XLite SIP UA + Asterisk
- Creating Asterisk accounts with a simple dial plan
- Configuration of XLite SIP UA (DTMF, codecs, NAT, RTP, timers, registration) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
- Registration, initiating, and receiving calls
- P2P calls using Linphone
- Analysis of SIP signalling using Wireshark
- Server configuration
- Capturing SIP signalling and RTP media streams
- SIP packet analysis and retrieval of specific calls
- Voice quality issues: Jitter buffer, DTMF signalling retrieval (RFC 2833, INFO), and troubleshooting codecs/DTMF (transcoding, GSM codec failure, DTMF tone duplication)
- VoIP monitoring techniques
- SDP, Instant Messaging, and Presence (IM&P)
- SDP parameters and attributes
- SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
- Practical exercises in IM&P with XLite and Linphone
- SIP call flows
- SIP Registration with DNS
- SIP SRV record implementation
- Registering SIP phones using DNS-SRV
- Call Flows with DNS
- Analyzing SIP call signalling using Wireshark
- Troubleshooting DNS timeouts and latency
- SIP Registration with DNS
- SIP trunks
- Establishing a test SIP trunk
- Troubleshooting issues (DoS, DDoS, fraud, CPS)
- SIP security issues
- Securing SIP with IPSec
- Implementing Secure SIP for security
- IP telephony fraud risks
- Preventing DDoS and other types of attacks
- Launching SIP-based VoIP services
- Switch configuration
- SIP client configuration and registration
- Software components
- Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
- Linux CentOS
- TDM2IP drivers
- Softphones (XLite, Linphone)
- Hardware components
- Server infrastructure
- TDM2IP card/gateway
- Hardphones (Polycom, Gigaset, Yealink)
- Softphone/Hardphone configuration
- Configuration settings
- Codecs
- User/Password/SIP Server/Proxy/Ports
- Operation and signalling for:
- 3-Way Calling
- Call Forwarding
- Attendant Call Transfer
- MWI, BLF
- Yealink autoprovisioning
- Vendor-dependent constraints
- Configuration settings
- SIP & Network Address Translation (NAT) challenges
- NAT types and structure
- STUN (Simple Traversal of UDP Through NATs)
- Quality of VoIP calls – troubleshooting
- Call connected but media missing
- Key QoS factors
- Delay, jitter, and play buffer size
- VoIP quality metrics
- RTCP – delay and jitter analysis
- MOS according to ITU-T G.107 E-model
- VoIP quality monitoring tools (Voipmonitor)
- Cloud-based IP telephony
- Wrap-up and addressing SIP and VoIP-related issues submitted by participants
Testimonials (7)
Good contact with the trainer
Radziszewski - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
Detailed analysis of the provided traces
Krzysztof - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
examples presented
Slawomir - NBP
Course - SIP protocol in VoIP
The instructor patient to explain and show the step on each lab session.
CHEE WAI KAN - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
Trainer can explain and give example for questions asked
SHAMSANI ABU BAKAR - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
Very informative and detail explanation
Syahrifendi Lukmanilhakim - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
The tools and the contents of the slide.