Get in Touch
 Duration 21 hours

Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. Variations 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 model
  3. Server operations
    1. Registration processes
    2. SIP servers operating in Proxy and Redirect modes
    3. Stateless versus stateful Proxy servers
    4. Location server functions
    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 analysis
    2. Request types
    3. Response types
    4. Demonstration of a call flow
    5. Header and parameter details
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Media description standards
    2. Standard codec lists
    3. Session negotiation protocols
  6. Call flows – SIP signaling
    1. SIP session – primary RFC 3261 example
    2. Sample call scenarios
    3. Conferencing and IP PBX integration
    4. Media changes during active sessions
    5. Utilizing IMP features
  7. Routing of SIP requests and responses
    1. VIA header mechanisms
    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 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 techniques
    2. Quality of Service (QoS)
    3. Integration of SIP and SDP in 3GPP IMS architecture
  11. Summary and discussion

Part II: Hands-on Training

  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. Managing registration, incoming, and outgoing calls
    4. Peer-to-peer calls using Linphone
  2. Analyzing SIP signaling using Wireshark
    1. Server configuration setup
    2. Registering SIP signaling and RTP media streams
    3. SIP packet analysis and specific call retrieval
    4. Addressing voice quality issues, jitter buffers, and DTMF signaling retrieval (RFC 2833, INFO). Troubleshooting codec and DTMF issues (transcoding, GSM codec failure, DTMF tone duplication)
    5. Implementing VoIP monitoring
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. SDP parameters and attributes
    2. SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practical exercises with IM&P using XLite and Linphone
  4. SIP call flows
    1. SIP Registration with DNS
      • SIP SRV record usage
      • Registering SIP phones using DNS-SRV
    2. Call flows with DNS
      • Analyzing SIP call signaling via 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. Securing SIP with IPSec
    2. Implementing security via Secure SIP
    3. Assessing fraud risks in IP telephony
    4. Mitigating DDOS and other types of attacks
  7. Deploying SIP-based VoIP services
    1. Switch configuration
    2. SIP client setup and registration
    3. Software components
      • Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
      • Linux CentOS environment
      • TDM2IP drivers
      • Softphones (XLite, Linphone)
    4. Hardware components
      • Server infrastructure
      • TDM2IP card/gateway
      • Hardphones (Polycom, Gigaset, Yealink)
  8. Softphone/Hardphone management
    1. Configuration settings
      • Codec selection
      • User/Password/SIP Server/Proxy/Ports
    2. Operations 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. NAT types and structures
    2. STUN (Simple Traversal of UDP Through NATs)
  10. VoIP call quality – troubleshooting
    1. Diagnosing connected calls with missing media
    2. Key QoS factors
      • Delay, jitter, and play buffer size
    3. VoIP quality metrics
      • RTCP – delay and jitter analysis
      • 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

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