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Course Outline
Part I: Introduction
- Introduction
- Historical context and drivers<\/li>
- VoIP types and evolutionary trends<\/li>
- Core SIP concepts<\/li>
- SIP standardization (RFC 3261 and associated standards)<\/li><\/ol><\/li>
- System Architecture
- User Agent (UA)<\/li>
- Predefined servers: Registrar, Location, Proxy, and Redirect<\/li>
- Application servers<\/li>
- User identification and addressing mechanisms<\/li>
- The SIP trapezoid model<\/li><\/ol><\/li>
- Server Operations
- Registration processes<\/li>
- SIP server behavior in Proxy and Redirect modes<\/li>
- Stateless versus stateful Proxy servers<\/li>
- Location server functions<\/li>
- SRV records and DNS integration<\/li>
- uri/url/urn, ENUM, and NAPTR records<\/li><\/ol><\/li>
- SIP Signaling Messages (including Instant Messaging & Presence – IMP extensions)
- Message anatomy<\/li>
- Request types<\/li>
- Response types<\/li>
- Call establishment examples<\/li>
- Header fields and parameters<\/li>
- IMP models<\/li><\/ol><\/li>
- SDP (Session Description Protocol)
- Media description formats<\/li>
- Standard codec lists<\/li>
- Session negotiation rules<\/li><\/ol><\/li>
- Call Flows and SIP Signaling
- SIP session mechanics – RFC 3261 examples<\/li>
- Representative call scenarios<\/li>
- Conferencing and IP PBX integration<\/li>
- Media modification during active sessions<\/li>
- Utilization of IMP<\/li><\/ol><\/li>
- Routing of SIP Requests and Responses
- VIA header<\/li>
- ROUTE and RECORD-ROUTE headers<\/li><\/ol><\/li>
- SIP-PSTN Interworking
- SIP-T and SIP-I protocols<\/li>
- SIP early media and SIP trunking<\/li>
- SIP-PSTN signaling interactions<\/li><\/ol><\/li>
- SIP Security Challenges
- Secure SIP, Secure RTP, and Secure RTCP<\/li>
- Common implementations of Secure SIP<\/li><\/ol><\/li>
- Operational Challenges and Future Directions
- NAT and firewall traversal techniques<\/li>
- Quality of Service (QoS)<\/li>
- SIP and SDP within the 3GPP IMS architecture<\/li><\/ol><\/li>
- Summary and Discussion<\/li><\/ol>
Part II: Hands-On Laboratory<\/p>
- SIP in a LAN Environment: XLite SIP UA + Asterisk
- Creating Asterisk accounts with a basic dial plan<\/li>
- Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)<\/li>
- Registration, call initiation, and call reception<\/li>
- Peer-to-peer calls using Linphone<\/li><\/ol><\/li>
- Analyzing SIP Signaling with Wireshark
- Server configuration<\/li>
- Capturing SIP signaling and RTP media streams<\/li>
- SIP packet analysis and specific call retrieval<\/li>
- Diagnosing voice quality issues: Jitter buffer, DTMF signaling retrieval (RFC 2833, INFO), and troubleshooting codec/DTMF issues (transcoding, GSM codec failures, DTMF tone duplication)<\/li>
- Utilizing VoIP monitor tools<\/li><\/ol><\/li>
- SDP, Instant Messaging, and Presence (IM&P)
- SDP parameters and attributes<\/li>
- SUBSCRIBE, PUBLISH, and MESSAGE SIP methods<\/li>
- Practical IM&P exercises with XLite and Linphone<\/li><\/ol><\/li>
- SIP Call Flows
- SIP Registration via DNS
- SIP SRV records<\/li>
- SIP phone registration using DNS-SRV<\/li><\/ul><\/li>
- Call Flows involving DNS
- Analyzing SIP call signaling with Wireshark<\/li>
- Troubleshooting DNS timeouts and latency<\/li><\/ul><\/li><\/ol><\/li>
- SIP Trunks
- Establishing a test SIP trunk<\/li>
- Troubleshooting (DOS, DDOS, fraud, cps)<\/li><\/ol><\/li>
- SIP Security Considerations
- SIP security using IPSec<\/li>
- Security via Secure SIP<\/li>
- IP telephony fraud risks<\/li>
- Prevention of DDOS and other attack vectors<\/li><\/ol><\/li>
- Deploying SIP-based VoIP Services
- Switch configuration<\/li>
- SIP client configuration and registration<\/li>
- Software Components
- Asterisk PBX / Freeswitch softswitch / Cisco Call Manager<\/li>
- Linux CentOS<\/li>
- TDM2IP drivers<\/li>
- Softphones (XLite, Linphone)<\/li><\/ul><\/li>
- Hardware Components
- Server infrastructure<\/li>
- TDM2IP card/gateway<\/li>
- Hardphones (Polycom, Gigaset, Yealink)<\/li><\/ul><\/li><\/ol><\/li>
- Softphone/Hardphone Management
- Configuration
- Codecs<\/li>
- User/Password/SIP Server/Proxy/Ports<\/li><\/ul><\/li>
- Operation and signaling for:
- 3-Way Calling<\/li>
- Call Forwarding<\/li>
- Attendant Call Transfer<\/li>
- MWI, BLF<\/li><\/ul><\/li>
- Yealink autoprovisioning<\/li>
- Vendor-specific constraints<\/li><\/ol><\/li>
- SIP & Network Address Translation (NAT) Issues
- NAT types and structures<\/li>
- STUN (Simple Traversal of UDP Through NATs)<\/li><\/ol><\/li>
- VoIP Call Quality – Troubleshooting
- Call connected but missing media<\/li>
- Key QoS factors
- Delay, jitter, play buffer size<\/li><\/ul><\/li>
- VoIP quality metrics
- RTCP – delay and jitter<\/li>
- MOS based on ITU-T G.107 E-model<\/li><\/ul><\/li>
- VoIP quality monitoring tools (Voipmonitor)<\/li><\/ol><\/li>
- Cloud-based IP Telephony<\/li>
- Wrap-up and discussion of SIP/VoIP issues submitted by participants<\/li><\/ol>
- Configuration
- SIP Registration via DNS
- SIP in a LAN Environment: XLite SIP UA + Asterisk
21 Hours
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.
