Get in Touch
 Duration 21 hours

Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and underlying motivations
    2. Evolution and various types of VoIP
    3. Core concepts of SIP
    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 model
  3. Server operations
    1. Registration processes
    2. SIP server functions in Proxy and Redirect modes
    3. Stateless versus stateful Proxy servers
    4. Location server functionalities
    5. SRV records and DNS interactions
    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. Illustrative call examples
    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 signaling
    1. SIP session – primary RFC 3261 example
    2. Sample call scenarios
    3. Conferencing and IP PBX integration
    4. Media modification during active sessions
    5. Utilization of 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 protocols
    2. SIP early media and SIP trunking
    3. SIP-PSTN signaling
  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
    2. Quality of Service (QoS)
    3. SIP and SDP within 3GPP IMS architecture
  11. Summary and discussion

Part II: Hands on

  1. SIP in LAN environments: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with basic dial plans
    2. Configuration of 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 with Wireshark
    1. Server configuration
    2. Registration of SIP signalling and RTP media streams
    3. SIP packet analysis. Retrieving specific calls
    4. Voice quality issues. Jitter buffer. Retrieval of DTMF signalling (RFC 2833, INFO). Codec and DTMF troubleshooting (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 record
      • SIP phone registration via DNS-SRV
    2. Call Flows with DNS
      • Analysing 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 with IPSec
    2. Security with Secure SIP
    3. IP telephony – fraud risks
    4. Preventing DDOS and other attack types
  7. Launching SIP-based VoIP services
    1. Configuration of 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
      • Hardphone (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-specific constraints
  9. SIP & Network Adress Translation (NAT) issues
    1. Type 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 related issues submitted by participants

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

2026-10-29 09:30
21 hours
Thessaloniki
911 EUR (Online)
911 EUR (Classroom)

SIP protocol in VoIP

2026-11-12 09:30
21 hours
Nicosia
911 EUR (Online)
911 EUR (Classroom)

SIP protocol in VoIP

2026-11-26 09:30
21 hours
Nicosia
911 EUR (Online)
1511 EUR (Classroom)

SIP protocol in VoIP

2026-12-10 09:30
21 hours
Athens
911 EUR (Online)
1511 EUR (Classroom)

SIP protocol in VoIP

2026-12-24 09:30
21 hours
Kaminia
911 EUR (Online)
1511 EUR (Classroom)

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