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Duration 21 hours
Course Outline
Part I: Theoretical Foundations
- Introduction
- Historical context and motivational drivers
- Types of VoIP and their evolutionary trajectory
- Core concepts of SIP
- SIP standardization (RFC 3261 and associated standards)
- Architecture
- User Agent (UA) components
- Predefined server roles: Registrar, Location, Proxy, and Redirect
- Application server functionalities
- Identification and addressing mechanisms
- The SIP trapezoid model
- Server Operations
- Registration processes
- SIP server behavior in Proxy and Redirect modes
- Distinctions between Stateless and Stateful Proxy servers
- Location server functions
- SRV records and DNS integration
- uri/url/urn, ENUM, and NAPTR records
- SIP Signalling Messages (including Instant Messaging & Presence – IMP extensions)
- Message structure breakdown
- Request formats
- Response types
- Illustrative call examples
- Headers and parameter details
- IMP models and applications
- SDP (Session Description Protocol)
- Media description standards
- Standard codec listings
- Session negotiation rules
- Call Flows – SIP Signalling
- SIP session overview with RFC 3261 reference
- Sample call scenarios
- Conferencing and IP PBX interactions
- Media switching during active sessions
- Implementation of IMP
- Routing of SIP Requests and Responses
- VIA header analysis
- ROUTE and RECORD-ROUTE header usage
- SIP-PSTN Interworking
- SIP-T and SIP-I protocols
- SIP early media and SIP trunking
- SIP-PSTN signalling mechanics
- SIP Security Challenges
- Secure SIP, Secure RTP, and Secure RTCP
- Common implementations of Secure SIP
- Practical Challenges and Future Perspectives
- NAT and firewall traversal techniques
- Quality of Service (QoS)
- SIP and SDP within 3GPP IMS architecture
- Summary and Discussion
Part II: Practical Application
- SIP in a LAN Environment: XLite SIP UA + Asterisk
- Creating Asterisk accounts with basic dial plans
- Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
- Managing registration, call initiation, and reception
- Peer-to-peer (P2P) calls using Linphone
- Analysis of SIP Signalling using Wireshark
- Server configuration for analysis
- Capture of SIP signalling and RTP media streams
- SIP packet inspection and specific call retrieval
- Diagnosing voice quality issues, jitter buffers, and DTMF signalling (RFC 2833, INFO). Troubleshooting codecs and DTMF (transcoding, GSM codec failure, tone duplication)
- Utilizing VoIP monitoring tools
- SDP, Instant Messaging, and Presence (IM&P)
- SDP parameters and attribute details
- SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
- Hands-on practice of IM&P with XLite and Linphone
- SIP Call Flows
- SIP Registration via DNS
- SIP SRV record structure
- Registering SIP phones using DNS-SRV
- Call Flows involving DNS
- Analysis of SIP call signalling via Wireshark
- Troubleshooting DNS timeouts and latency issues
- SIP Registration via DNS
- SIP Trunks
- Establishing test SIP trunks
- Troubleshooting DOS, DDOS, fraud, and CPS issues
- SIP Security Issues
- Implementing SIP security with IPSec
- Security configurations with Secure SIP
- Assessing fraud risks in IP telephony
- Mitigating DDOS and other attack vectors
- Deploying SIP-based VoIP Services
- Switch configuration
- SIP client setup and registration
- Software Components
- Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
- Linux CentOS environment
- TDM2IP drivers
- Softphones (XLite, Linphone)
- Hardware Components
- Server infrastructure
- TDM2IP card/gateway
- Hardphones (Polycom, Gigaset, Yealink)
- Softphone/Hardphone Management
- Configuration Settings
- Codec selection
- User/Password/SIP Server/Proxy/Ports
- Operational Signalling for:
- 3-Way Calling
- Call Forwarding
- Attendant Call Transfer
- MWI, BLF
- Yealink autoprovisioning procedures
- Addressing vendor-specific constraints
- Configuration Settings
- SIP & Network Address Translation (NAT) Challenges
- NAT types and structural variations
- STUN (Simple Traversal of UDP Through NATs)
- VoIP Call Quality – Troubleshooting
- Diagnosing connected calls with missing media
- Key QoS Factors
- Delay, jitter, and play buffer size
- VoIP Quality Metrics
- RTCP – delay and jitter analysis
- MOS calculation based on ITU-T G.107 E-model
- Monitoring VoIP quality with tools (Voipmonitor)
- Cloud-based IP Telephony Solutions
- Final Wrap-up and Review of Participant-Submitted SIP and VoIP Issues
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.