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Course Outline

 Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. Variations of VoIP and its development
    3. Core SIP concepts
    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. SIP trapezoid
  3. Servers and their operation
    1. Registration processes
    2.  SIP server operation 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 signaling 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 codec list
    3. Session negotiation rules
  6. Call flows – SIP signaling
    1. SIP session – primary example from RFC 3261
    2. Sample call scenarios
    3. Conferencing and IP PBX
    4. Media changes during a session
    5. Utilizing IMP
  7. Routing of SIP requests and responses
    1. VIA header
    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 signaling
  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 and firewall traversal
    2. QoS
    3. SIP and SDP within 3GPP IMS architecture
  11. Wrap-up and discussion

Part II: Hands on

  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. Analysis of SIP signaling using Wireshark
    1. Server configuration
    2. Registration of SIP signaling and RTP media streams
    3. SIP packet analysis. Isolating a specific call
    4. Voice quality issues. Jitter buffer. Extracting DTMF signaling (RFC 2833, INFO). Troubleshooting codec and DTMF issues (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 with DNS
      • SIP SRV record
      • SIP phone registration using DNS-SRV
    2. Call Flows with DNS
      • Analysis of SIP call signaling using Wireshark
      • Troubleshooting – DNS timeout, latency
  5. SIP trunks
    1. Establishing a test SIP trunk
    2. Troubleshooting (DOS, DDOS, fraud, cps)
  6. SIP security concerns
    1. SIP security with IPSec
    2. Security using Secure SIP
    3. IP telephony – fraud risks
    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
    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 structure of NATs
    2. STUN (Simple Traversal of UDP Through NATs)
  10. Quality of VoIP calls – troubleshooting
    1. Call connected – missing media
    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

2026-10-16 09:30
21 hours
Cluj-Napoca, Pitești Street
743 EUR (Online)
1343 EUR (Classroom)

SIP protocol in VoIP

2026-10-30 09:30
21 hours
Timisoara, Iosif Bulbuca Boulevard
743 EUR (Online)
1343 EUR (Classroom)

SIP protocol in VoIP

2026-11-13 09:30
21 hours
Constanta, Mircea cel Batran Street
743 EUR (Online)
1343 EUR (Classroom)

SIP protocol in VoIP

2026-11-27 09:30
21 hours
Iasi, City Center
743 EUR (Online)
1343 EUR (Classroom)

SIP protocol in VoIP

2026-12-11 09:30
21 hours
Bucharest, City Center
743 EUR (Online)
1343 EUR (Classroom)

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