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Course Outline

 Part I: Introduction

  1. Introduction
    1. Historical context and motivational drivers
    2. VoIP categories and evolutionary development
    3. Core SIP concepts
    4. SIP standardization frameworks (including RFC 3261 and related standards)
  2. Architecture
    1. UA – User Agent roles
    2. Predefined server types: Registrar, Location, Proxy, and Redirect
    3. Application server functions
    4. Identification and addressing mechanisms
    5. The SIP trapezoid model
  3. Server Operations
    1. Registration processes
    2.  SIP server functionality in Proxy and Redirect modes
    3. Stateless versus stateful Proxy server implementations
    4. Location server operations
    5. SRV records and DNS integration
    6. uri/url/urn, ENUM, and NAPTR record usage
  4. SIP Signalling Messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure analysis
    2. Request types
    3. Response types
    4. Call example walkthrough
    5. Header and parameter details
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Media description methods
    2. Standard codec lists
    3. Session negotiation rules
  6. Call Flows – SIP Signalling
    1. SIP session – primary  RFC 3261 example
    2. Representative call scenarios
    3. Conferencing and IP PBX integration
    4. Media changes during active sessions
    5. Utilizing IMP features
  7. Routing of SIP Requests and Responses
    1. VIA header analysis
    2. ROUTE and RECORD-ROUTE header functions
  8. SIP-PSTN Interworking
    1. SIP-T and SIP-I protocols
    2. SIP early media and SIP trunking
    3. SIP-PSTN signalling interactions
  9. SIP – Security Challenges
    1. Secure SIP, Secure RTP, and Secure RTCP
    2. Common Secure SIP implementations
  10. Practical Challenges and Future Perspectives
    1. NAT and firewall traversal techniques
    2. QoS management
    3. SIP and SDP within 3GPP IMS architecture
  11. Wrap-up and Discussion

Part II: Hands On

  1. SIP in LAN Environments: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with basic dial plans
    2. Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
    3. Handling registration, initiating, and receiving calls
    4. P2P calls using Linphone
  2. Analyzing SIP Signalling with Wireshark
    1. Server configuration
    2. Capturing SIP signalling and RTP media streams
    3. SIP packet analysis and specific call retrieval
    4. Voice quality issues, jitter buffers, and DTMF signalling retrieval (RFC 2833, INFO). Troubleshooting codecs and DTMF (transcoding, GSM codec failure, DTMF tone duplication)
    5. VoIP monitoring
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. SDP parameters and attributes
    2. SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practising IM&P with XLite and Linphone
  4. SIP Call Flows
    1. SIP Registration with DNS
      • SIP SRV records
      • SIP phone registration using DNS-SRV
    2. Call Flows with DNS
      • Analyzing SIP call signalling using Wireshark
      • Troubleshooting DNS timeout and latency
  5. SIP Trunks
    1. Establishing a test SIP trunk
    2. Troubleshooting (DOS, DDOS, fraud, cps)
  6. SIP Security Issues
    1. SIP security via IPSec
    2. Security with Secure SIP
    3. IP telephony – fraud risk assessment
    4. Preventing DDOS and other attack types
  7. Launching SIP-Based VoIP Services
    1. Switch configuration
    2. SIP client configuration and registration
    3. Software
      • Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
      • Linux CentOS
      • TDM2IP drivers
      • Softphones (XLite, Linphone)
    4. Hardware
      • Server infrastructure
      • TDM2IP card/gateway
      • Hardphones (Polycom, Gigaset, Yealink)
  8. Softphone/Hardphone Management
    1. Configuration
      • Codecs
      • User/Password/SIP Server/Proxy/Ports
    2. Operation and signalling 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 structures
    2. STUN (Simple Traversal of UDP Through NATs)
  10. VoIP Call Quality – Troubleshooting
    1. Call connected but missing media
    2. Key QoS factors
      • Delay, jitter, and play buffer size
    3. VoIP quality metrics
      • RTCP – delay and jitter analysis
      • 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

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SIP protocol in VoIP

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SIP protocol in VoIP

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