Get in Touch
 Duration 21 hours (3 days)

Course Outline

 Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. Evolution and types of VoIP
    3. Key concepts of SIP
    4. SIP standardization (RFC 3261 and related standards)
  2. Architecture
    1. UA – User Agent
    2. Standard servers: Registrar, Location, Proxy, and Redirect
    3. Application servers
    4. User identification and addressing schemes
    5. The SIP trapezoid model
  3. Server Operations
    1. Registration processes
    2. SIP server behavior in Proxy and Redirect modes
    3. Stateless versus stateful Proxy servers
    4. Location server functionality
    5. SRV records and DNS integration
    6. uri/url/urn, ENUM, and NAPTR records
  4. SIP Signalling Messages (including Instant Messaging & Presence – IMP extensions)
    1. Message anatomy
    2. Request methods
    3. Response codes
    4. Case study: Call establishment
    5. Headers and parameter usage
    6. IMP data models
  5. SDP (Session Description Protocol)
    1. Media description mechanisms
    2. Standard codec listings
    3. Rules for session negotiation
  6. Call Flows – SIP Signalling
    1. SIP session lifecycle – Key RFC 3261 examples
    2. Various call scenario walkthroughs
    3. Conferencing and IP PBX integration
    4. Media modification during an active session
    5. Utilization of IMP
  7. Routing of SIP Requests and Responses
    1. VIA header mechanics
    2. ROUTE and RECORD-ROUTE header usage
  8. SIP-PSTN Interworking
    1. SIP-T and SIP-I protocols
    2. SIP early media and trunking
    3. Signalling between SIP and PSTN networks
  9. SIP Security Challenges
    1. Secure SIP, Secure RTP, and Secure RTCP
    2. Common implementations of Secure SIP
  10. Practical Challenges and Future Perspectives
    1. NAT traversal and firewall considerations
    2. Quality of Service (QoS)
    3. Role of SIP and SDP in 3GPP IMS architecture
  11. Summary and Open Discussion

Part II: Hands-on Practical

  1. SIP in a LAN Environment: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with basic dial plans
    2. Configuring XLite SIP UA (DTMF, codecs, NAT, RTP, timers, registration) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
    3. Handling registration, call initiation, and call reception
    4. Peer-to-peer (P2P) calls using Linphone
  2. Analyzing SIP Signalling with Wireshark
    1. Server setup and configuration
    2. Capturing SIP signalling and RTP media streams
    3. SIP packet analysis and retrieving specific call details
    4. Diagnosing voice quality issues, jitter buffers, and DTMF signalling retrieval (RFC 2833, INFO); troubleshooting codecs and DTMF (transcoding, GSM codec failures, DTMF tone duplication)
    5. Implementing a VoIP monitor
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. SDP parameters and attribute management
    2. Using SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practicing IM&P with XLite and Linphone
  4. SIP Call Flows
    1. SIP Registration via DNS
      • Understanding SIP SRV records
      • Registering SIP phones using DNS-SRV
    2. Call Flows involving DNS
      • Analysing SIP call signalling with Wireshark
      • Troubleshooting DNS timeouts and latency issues
  5. SIP Trunks
    1. Establishing a test SIP trunk
    2. Troubleshooting DOS, DDOS, fraud, and CPS issues
  6. SIP Security Issues
    1. Implementing SIP security with IPSec
    2. Securing communications with Secure SIP
    3. Risks of fraud in IP telephony
    4. Mitigating DDOS and other attack vectors
  7. Launching SIP-Based VoIP Services
    1. Configuring the switching system
    2. SIP client setup and registration
    3. Software Stack
      • Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
      • Linux CentOS operating system
      • TDM2IP drivers
      • Softphones (XLite, Linphone)
    4. Hardware Requirements
      • Server infrastructure
      • TDM2IP card/gateway
      • Hardphones (Polycom, Gigaset, Yealink)
  8. Softphone/Hardphone Management
    1. Configuration Settings
      • Codec selection
      • User credentials, SIP Server, Proxy, and Port settings
    2. Operation and signalling for:
      • 3-Way Calling
      • Call Forwarding
      • Attendant Call Transfer
      • Message Waiting Indicator (MWI) and Busy Lamp Field (BLF)
    3. Yealink auto-provisioning
    4. Handling vendor-specific constraints
  9. SIP & Network Address Translation (NAT) Challenges
    1. NAT types and structural considerations
    2. STUN (Simple Traversal of UDP Through NATs)
  10. VoIP Call Quality – Troubleshooting
    1. Scenario: Call connected but media missing
    2. Key QoS factors
      • Delay, jitter, and play buffer size
    3. VoIP Quality Metrics
      • RTCP – measuring delay and jitter
      • MOS scoring based on ITU-T G.107 E-model
    4. Using VoIP quality monitoring tools (Voipmonitor)
  11. Cloud-Based IP Telephony
  12. Wrap-up and addressing participant-submitted SIP and VoIP issues

Need help picking the right course?
southafrica@nobleprog.co.za or +27 (0)10 005 5793

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.

Kak Khen Man - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP

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