Get in Touch
 Duration 21 hours (3 days)

Course Outline

Part I: Introduction

  1. Introduction
    1. Historical context and motivation
    2. Evolution and types of VoIP
    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. Servers and their operational roles
    1. Registration processes
    2. SIP server operations 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 signalling messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure anatomy
    2. Request mechanisms
    3. Response handling
    4. Practical example of a call flow
    5. Headers and parameter details
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Media description standards
    2. Standard list of supported codecs
    3. Session negotiation rules
  6. Call flows – SIP signalling
    1. SIP session – main RFC 3261 example
    2. Sample call scenarios
    3. Conferencing and IP PBX integration
    4. Dynamic media changes during a session
    5. Utilizing IMP features
  7. Routing of SIP requests and responses
    1. VIA header analysis
    2. ROUTE and RECORD-ROUTE header usage
  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 challenges
    1. Secure SIP, Secure RTP, and Secure RTCP
    2. Typical implementations of Secure SIP
  10. Practical challenges and future perspectives
    1. NAT and firewall traversal techniques
    2. Quality of Service (QoS)
    3. SIP and SDP within the 3GPP IMS architecture
  11. Wrap-up and discussion

Part II: Hands-on Lab

  1. SIP in a LAN environment: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with a basic dial plan
    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. Peer-to-peer (P2P) calls using Linphone
  2. Analyzing SIP signalling using Wireshark
    1. Server configuration setup
    2. Capturing SIP signalling and RTP media streams
    3. SIP packet analysis. Retrieval of a specific call record
    4. Voice quality issues. Jitter buffer management. Retrieval of DTMF signalling (RFC 2833, INFO). Codec and DTMF troubleshooting (transcoding, GSM codec failure, DTMF tone duplication)
    5. VoIP monitoring tools
  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 configuration
      • SIP phone registration using DNS-SRV
    2. Call Flows with DNS
      • Analysing SIP call signalling using 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 issues
    1. SIP security with IPSec
    2. Security with Secure SIP
    3. IP telephony – risk of frauds
    4. Preventing DDOS and other types of attacks
  7. Launching SIP-based VoIP services
    1. Configuration of a switch
    2. SIP client configuration and registration
    3. Software stack
      • Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
      • Linux CentOS
      • TDM2IP drivers
      • Softphones (XLite, Linphone)
    4. Hardware components
      • Server infrastructure
      • 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) problems
    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

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