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Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. VoIP types and evolutionary trends
    3. Core SIP concepts
    4. SIP standardization (RFC 3261 and associated standards)
  2. Architecture
    1. UA – User Agent
    2. Predefined servers: Registrar, Location, Proxy, and Redirect
    3. Application servers
    4. Identification and addressing mechanisms
    5. The SIP trapezoid
  3. Servers and operational mechanics
    1. Registration processes
    2. SIP servers operating in Proxy and Redirect modes
    3. Stateless versus stateful Proxy servers
    4. Location servers
    5. SRV records and DNS integration
    6. uri/url/urn, ENUM, and NAPTR records
  4. SIP signaling messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure analysis
    2. Request handling
    3. Response mechanisms
    4. Call example walkthroughs
    5. Headers and parameter definitions
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Media description formats
    2. Standard codec lists
    3. Session negotiation rules
  6. Call flows – SIP signaling
    1. SIP session overview – primary RFC 3261 example
    2. Sample call scenario analysis
    3. Conferencing and IP PBX integration
    4. Media modification during sessions
    5. Utilization of IMP
  7. Routing of SIP requests and responses
    1. VIA header usage
    2. ROUTE and RECORD-ROUTE headers
  8. SIP-PSTN interworking
    1. SIP-T and SIP-I protocols
    2. SIP early media and trunking
    3. SIP-PSTN signaling interactions
  9. SIP security considerations
    1. Secure SIP, Secure RTP, and Secure RTCP
    2. Common implementations of Secure SIP
  10. Practical challenges and future perspectives
    1. NAT and firewall traversal techniques
    2. Quality of Service (QoS)
    3. Integration of SIP and SDP in 3GPP IMS architecture
  11. Summary and discussion

Part II: Hands-On

  1. SIP in a LAN environment: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with a simple dial plan
    2. Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
    3. Registration, call initiation, and reception
    4. P2P calls using Linphone
  2. Analyzing SIP signaling using Wireshark
    1. Server configuration setup
    2. Registration of SIP signaling and RTP media streams
    3. SIP packet analysis and retrieval of specific calls
    4. Voice quality issues: Jitter buffer, DTMF signaling retrieval (RFC 2833, INFO), and troubleshooting codecs/DTMF (transcoding, GSM codec failure, DTMF tone duplication)
    5. VoIP monitoring tools
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. SDP parameters and attributes
    2. SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practical IM&P exercises with XLite and Linphone
  4. SIP call flows
    1. SIP Registration with DNS
      • SIP SRV record implementation
      • SIP phone registration via DNS-SRV
    2. Call Flows with DNS
      • Analysis of SIP call signaling using Wireshark
      • Troubleshooting DNS timeouts and latency
  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 SIP communications
    3. Risks of fraud in IP telephony
    4. Prevention of DDOS and other attack vectors
  7. Launching SIP-based VoIP services
    1. Switch configuration
    2. SIP client configuration and registration
    3. Software stack
      • Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
      • Linux CentOS
      • TDM2IP drivers
      • Softphones (XLite, Linphone)
    4. Hardware requirements
      • Server infrastructure
      • TDM2IP card/gateway
      • Hardphones (Polycom, Gigaset, Yealink)
  8. Softphone/Hardphone management
    1. Configuration settings
      • Codecs
      • User/Password/SIP Server/Proxy/Ports
    2. Operation and signaling for:
      • 3-Way Calling
      • Call Forwarding
      • Attendant Call Transfer
      • MWI, BLF
    3. Yealink autoprovisioning
    4. Vendor-specific constraints
  9. SIP & Network Address Translation (NAT) challenges
    1. NAT types and structure
    2. STUN (Simple Traversal of UDP Through NATs)
  10. VoIP call quality – troubleshooting
    1. Diagnosing connected calls with missing media
    2. Key QoS factors
      • Delay, jitter, and play buffer size
    3. VoIP quality metrics
      • RTCP for delay and jitter measurement
      • MOS based on ITU-T G.107 E-model
    4. VoIP quality monitoring tools (Voipmonitor)
  11. Cloud-based IP telephony
  12. Wrap-up and addressing SIP and VoIP issues submitted by participants
 21 Hours

Number of participants


Price per participant

Open Training Courses require 5+ 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.

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

Upcoming Courses

SIP protocol in VoIP

2026-10-20 09:30
21 hours
Kuala Lumpur KL Sentral
21661 MYR (Online)
22636 MYR (Classroom)

SIP protocol in VoIP

2026-11-03 09:30
21 hours
Bayan Lepas, iDEAL
21661 MYR (Online)
22396 MYR (Classroom)

SIP protocol in VoIP

2026-11-17 09:30
21 hours
Kuala Lumpur KL Sentral
21661 MYR (Online)
22636 MYR (Classroom)

SIP protocol in VoIP

2026-12-01 09:30
21 hours
Bayan Lepas, iDEAL
21661 MYR (Online)
22396 MYR (Classroom)

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