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Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. Various types of VoIP and their evolution
    3. Core concepts of SIP
    4. SIP standardization (RFC 3261 and other relevant 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 their operation
    1. Registration processes
    2. SIP server functioning in Proxy and Redirect modes
    3. Stateless and stateful Proxy servers
    4. Location server functionality
    5. SRV records and DNS
    6. uri/url/urn, ENUM, and NAPTR records
  4. SIP signalling messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure
    2. Request types
    3. Response types
    4. Example call flow
    5. Headers and parameters
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Media description
    2. Standard list of codecs
    3. Session negotiation rules
  6. Call flows – SIP signalling
    1. SIP session – primary RFC 3261 example
    2. Sample call scenarios
    3. Conferencing and IP PBX integration
    4. Media changes during a session
    5. Utilizing 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
    2. SIP early media and SIP trunking
    3. SIP-PSTN signalling
  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 and firewall traversal
    2. Quality of Service (QoS)
    3. SIP and SDP within 3GPP IMS architecture
  11. Summary and discussion

Part II: Hands-on Labs

  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, initiating, and receiving calls
    4. P2P calls using Linphone
  2. Analyzing SIP signalling using Wireshark
    1. Server configuration
    2. Registration of SIP signalling and RTP media streams
    3. SIP packet analysis and retrieval of specific calls
    4. Voice quality issues: Jitter buffer, DTMF signalling retrieval (RFC 2833, INFO), and troubleshooting codec/DTMF issues (transcoding, GSM codec failure, DTMF tone duplication)
    5. Using VoIP monitor
  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 records
      • Registering SIP phones using DNS-SRV
    2. Call flows involving DNS
      • Analyzing SIP call signalling with Wireshark
      • Troubleshooting DNS timeouts and latency
  5. SIP trunks
    1. Establishing a test SIP trunk
    2. Troubleshooting issues (DOS, DDOS, fraud, CPS)
  6. SIP security issues
    1. Implementing SIP security with IPSec
    2. Securing with Secure SIP
    3. Risks of fraud in IP telephony
    4. Preventing DDOS and other types of attacks
  7. Launching SIP-based VoIP services
    1. Configuring a switch
    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
      • Hardphones (Polycom, Gigaset, Yealink)
  8. Softphone/Hardphone
    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-dependent constraints
  9. SIP & Network Address Translation (NAT) issues
    1. Types and structure of NAT
    2. STUN (Simple Traversal of UDP Through NATs)
  10. Quality of VoIP calls – troubleshooting
    1. Call connected but media missing
    2. Key QoS factors
      • Delay, jitter, 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 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

2026-10-21 09:30
21 hours
Prague, Prague Castle
4500 EUR (Online)
5100 EUR (Classroom)

SIP protocol in VoIP

2026-11-04 09:30
21 hours
Brno
4500 EUR (Online)
4500 EUR (Classroom)

SIP protocol in VoIP

2026-11-18 09:30
21 hours
Ostrava
4500 EUR (Online)
4500 EUR (Classroom)

SIP protocol in VoIP

2026-12-02 09:30
21 hours
Prague, Prague Castle
4500 EUR (Online)
5100 EUR (Classroom)

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