Get in Touch
 Duration 21 hours

Course Outline

 Part I: Introduction

  1. Introduction
    1. Historical context and drivers
    2. Variations of VoIP and its development
    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 functions and operations
    1. Registration processes
    2.  SIP server behavior 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, ENUM, and NAPTR records
  4. SIP signaling messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure
    2. Request types
    3. Response types
    4. Call examples
    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 RFC 3261 example
    2. Sample call scenarios
    3. Conferencing and IP PBX integration
    4. Media modification 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 interactions
  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 the 3GPP IMS architecture
  11. Summary 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 the 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. Capturing SIP signaling and RTP media streams
    3. SIP packet analysis and retrieving specific calls
    4. Voice quality issues. Jitter buffer. Extracting DTMF signaling (RFC 2833, INFO). Troubleshooting codecs and DTMF (transcoding, GSM codec failures, 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. Practical IM&P exercises 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
      • Analyzing SIP call signaling using Wireshark
      • Troubleshooting – DNS timeouts and latency
  5. SIP trunks
    1. Setting up a test SIP trunk
    2. Troubleshooting (DOS, DDOS, fraud, cps)
  6. SIP security considerations
    1. SIP security using IPSec
    2. Security through Secure SIP
    3. IP telephony – risk of fraud
    4. Preventing DDOS and other attack vectors
  7. Implementing 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) issues
    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, play buffer size
    3. VoIP quality metrics
      • RTCP – delay and jitter
      • MOS based on the 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 raised 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

2026-11-23 09:30
21 hours
Buenos Aires - Libertador
8730 USD (Online)
10230 USD (Classroom)

SIP protocol in VoIP

2026-12-07 09:30
21 hours
Còrdoba - Ecipsa Tower
8730 USD (Online)
10230 USD (Classroom)

SIP protocol in VoIP

2026-12-21 09:30
21 hours
Puerto Madero - Torre Colonos Sur
8730 USD (Online)
10230 USD (Classroom)

SIP protocol in VoIP

2027-01-04 09:30
21 hours
Buenos Aires - Laminar Catalinas
8730 USD (Online)
10230 USD (Classroom)

SIP protocol in VoIP

2027-01-18 09:30
21 hours
Buenos Aires - Libertador
8730 USD (Online)
10230 USD (Classroom)

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