Get in Touch
 Duration 21 hours

Course Outline

 Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. Varieties of VoIP and their evolution
    3. Core concepts of SIP
    4. SIP standardization (RFC 3261 and other pertinent standards)
  2. Architecture
    1. UA – User Agent
    2. Predefined server types: 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 functionality
    5. SRV records and DNS integration
    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. Modifying media during an active session
    5. Utilizing IMP features
  7. Routing of SIP requests and responses
    1. VIA header mechanics
    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 signalling interactions
  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 techniques
    2. QoS implementation
    3. SIP and SDP within the 3GPP IMS architecture
  11. Summary and discussion

Part II: Hands-on Practice

  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, call initiation, and reception
    4. P2P calls using Linphone
  2. Analysing SIP signalling using Wireshark
    1. Server configuration
    2. Capturing 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 codecs and DTMF (transcoding, GSM codec failure, DTMF tone duplication)
    5. VoIP monitor utilization
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. SDP parameters and attributes
    2. SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practical application of IM&P 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 with DNS
      • Analysing SIP call signalling via Wireshark
      • Troubleshooting – DNS timeout and latency issues
  5. SIP trunks
    1. Establishing a test SIP trunk
    2. Troubleshooting (DOS, DDOS, fraud, cps)
  6. SIP security concerns
    1. SIP security utilizing IPSec
    2. Security via Secure SIP
    3. IP telephony – risks of fraud
    4. Preventing DDOS and other forms 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 management
    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 Address Translation (NAT) challenges
    1. Types and structure of NATs
    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, and play buffer size
    3. VoIP quality metrics
      • RTCP – delay and jitter analysis
      • 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-related issues submitted by participants

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