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Duration 21 hours
Course Outline
Part I: Introduction
- Introduction
- Historical context and motivation
- Varieties of VoIP and their evolution
- Core concepts of SIP
- SIP standardization (RFC 3261 and other pertinent standards)
- Architecture
- UA – User Agent
- Predefined server types: Registrar, Location, Proxy, and Redirect
- Application servers
- Identification and addressing mechanisms
- The SIP trapezoid model
- Server operations
- Registration processes
- SIP servers operating in Proxy and Redirect modes
- Stateless versus 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 flow
- Headers and parameters
- IMP models
- SDP (Session Description Protocol)
- Media description
- Standard list of codecs
- Session negotiation rules
- Call flows – SIP signalling
- SIP session – primary RFC 3261 example
- Sample call scenarios
- Conferencing and IP PBX integration
- Modifying media during an active session
- Utilizing IMP features
- Routing of SIP requests and responses
- VIA header mechanics
- 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 considerations
- Secure SIP, Secure RTP, and Secure RTCP
- Common implementations of Secure SIP
- Practical challenges and future perspectives
- NAT and firewall traversal techniques
- QoS implementation
- SIP and SDP within the 3GPP IMS architecture
- Summary and discussion
Part II: Hands-on Practice
- SIP in a LAN environment: XLite SIP UA + Asterisk
- Creating Asterisk accounts with a basic dial plan
- Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
- Registration, call initiation, and reception
- P2P calls using Linphone
- Analysing 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 and DTMF (transcoding, GSM codec failure, DTMF tone duplication)
- VoIP monitor utilization
- SDP, Instant Messaging, and Presence (IM&P)
- SDP parameters and attributes
- SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
- Practical application of IM&P with XLite and Linphone
- SIP call flows
- SIP Registration with DNS
- SIP SRV records
- Registering SIP phones using DNS-SRV
- Call Flows with DNS
- Analysing SIP call signalling via Wireshark
- Troubleshooting – DNS timeout and latency issues
- SIP Registration with DNS
- SIP trunks
- Establishing a test SIP trunk
- Troubleshooting (DOS, DDOS, fraud, cps)
- SIP security concerns
- SIP security utilizing IPSec
- Security via Secure SIP
- IP telephony – risks of fraud
- Preventing DDOS and other forms of attacks
- Launching SIP-based VoIP services
- Switch configuration
- SIP client configuration and registration
- Software
- Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
- Linux CentOS
- TDM2IP drivers
- Softphones (XLite, Linphone)
- Hardware
- Server
- TDM2IP card/gateway
- Hardphone (Polycom, Gigaset, Yealink)
- Softphone/Hardphone management
- Configuration
- Codecs
- User/Password/SIP Server/Proxy/Ports
- Operation and signalling for:
- 3-Way Calling
- Call Forwarding
- Attendant Call Transfer
- MWI, BLF
- Yealink autoprovisioning
- Vendor-specific constraints
- Configuration
- SIP & Network Address Translation (NAT) challenges
- Types and structure of NATs
- STUN (Simple Traversal of UDP Through NATs)
- VoIP call quality – troubleshooting
- Call connected but missing media
- Key QoS factors
- Delay, jitter, and play buffer size
- VoIP quality metrics
- RTCP – delay and jitter analysis
- MOS based on the 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.