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Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and underlying motivations
    2. Variations of VoIP and their evolution
    3. Core concepts of SIP
    4. SIP standardization processes (RFC 3261 and associated standards)
  2. System Architecture
    1. User Agents (UA)
    2. Predefined server roles: Registrar, Location, Proxy, and Redirect
    3. Application server functions
    4. Identification mechanisms and addressing schemes
    5. The SIP trapezoid model
  3. Server Operations
    1. The registration process
    2. Operating SIP servers in Proxy and Redirect modes
    3. Differences between stateless and stateful Proxy servers
    4. Location server functionality
    5. SRV records and DNS integration
    6. uri/url/urn structures, ENUM, and NAPTR records
  4. SIP Signaling Messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure breakdown
    2. Request types
    3. Response types
    4. Illustrative call examples
    5. Header fields and parameters
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Media description
    2. Standard codec lists
    3. Session negotiation rules
  6. Call Flows – SIP Signaling
    1. SIP session basics – primary RFC 3261 examples
    2. Representative call scenarios
    3. Conferencing and IP PBX integration
    4. Media changes during an active session
    5. Utilizing IMP
  7. Routing SIP Requests and Responses
    1. The VIA header
    2. ROUTE and RECORD-ROUTE headers
  8. SIP-PSTN Interworking
    1. SIP-T and SIP-I protocols
    2. SIP early media and SIP trunking
    3. Signaling between SIP and PSTN
  9. SIP Security Challenges
    1. Secure SIP, Secure RTP, and Secure RTCP
    2. Common implementations of Secure SIP
  10. Practical Issues and Future Perspectives
    1. NAT traversal and firewall challenges
    2. Quality of Service (QoS)
    3. SIP and SDP within the 3GPP IMS architecture
  11. Conclusion and Discussion

Part II: Hands-On Practical Labs

  1. SIP in a LAN Environment: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with a basic dial plan
    2. Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
    3. Registration, initiating, and receiving calls
    4. P2P calls using Linphone
  2. Analyzing SIP Signaling with Wireshark
    1. Server configuration
    2. Registration of SIP signaling and RTP media streams
    3. SIP packet analysis and retrieval of specific calls
    4. Voice quality issues, Jitter buffer, and DTMF signaling 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. Practicing IM&P with XLite and Linphone
  4. SIP Call Flows
    1. SIP Registration using DNS
      • SIP SRV records
      • Registering SIP phones via DNS-SRV
    2. Call Flows involving DNS
      • Analyzing SIP call signaling using Wireshark
      • Troubleshooting – DNS timeouts and latency
  5. SIP Trunks
    1. Establishing a test SIP trunk
    2. Troubleshooting (DOS, DDOS, fraud, cps)
  6. SIP Security Issues
    1. SIP security using IPSec
    2. Security with Secure SIP
    3. IP telephony – risk of frauds
    4. Preventing DDOS and other types of attacks
  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
      • TDM2IP card/gateway
      • Hardphone (Polycom, Gigaset, Yealink)
  8. Softphone/Hardphone Management
    1. Configuration
      • 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. Types and structures of NATs
    2. STUN (Simple Traversal of UDP Through NATs)
  10. Voice Call Quality – Troubleshooting
    1. Call connected – missing media
    2. Key QoS factors
      • Delay, jitter, and play buffer size
    3. VoIP quality metrics
      • RTCP – delay and jitter
      • MOS according to 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-related 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

2026-10-06 09:30
21 hours
Tijuana - Business Centers
109,890 MXN (Online)
139,890 MXN (Classroom)

SIP protocol in VoIP

2026-10-20 09:30
21 hours
Leon - Torre San Mateo
109,890 MXN (Online)
139,890 MXN (Classroom)

SIP protocol in VoIP

2026-11-03 09:30
21 hours
Tuxtla Gutiérrez - Santa Elena
109,890 MXN (Online)
139,890 MXN (Classroom)

SIP protocol in VoIP

2026-11-17 09:30
21 hours
Ciudad Juárez - Bulevar Tomás Fernández
109,890 MXN (Online)
139,890 MXN (Classroom)

SIP protocol in VoIP

2026-12-01 09:30
21 hours
Toluca - La Merced
109,890 MXN (Online)
139,890 MXN (Classroom)

SIP protocol in VoIP

2026-12-15 09:30
21 hours
Toluca - Samara Shops Tower A
109,890 MXN (Online)
139,890 MXN (Classroom)

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