SIP protocol in VoIP Training Course

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Last updated

Course Code



21 hours (usually 3 days including breaks)


The course consists of two complementary parts – a theoretical and a practical one. The first is a one day introduction covering motivation, philosophy, fundamentals and rules of operation of the SIP protocol and ways it is used to implement telecom services with focus on IP telephony and VoIP. The second two-day part enables participants to learn practical aspects of service operation within a framework of hand-on laboratory exercises giving detailed insight into configuration of components of SIP telephony architecture, SIP signalling at both message sequence chart and internal message structure levels, and assists in understanding of typical problems and troubleshooting including security and telecom fraud aspects. The trainers will share their experience in launching, operation and management of SIP telephony covering also virtualization and cloud based solutions. Practical part is presented using both SIP hardphones and softphones and IP telephony servers (Asterisk and Freeswitch). Participants can take advantage of the fact trainers have got rich technical and business experience in IP telephony and submit their own problems and questions. They will be included in the agenda at wrap-up as a supplement to the training to meet current urgent needs of clients. Training is addressed to participants with basic knowledge and experience in telecom services – specifically in VoIP and IP networking.

Course Outline

 Part I: Introduction

  1. Introduction
    1. History and motivation
    2. Types of VoIP and its evolution
    3. SIP – main concepts
    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
    5. SIP trapezoid
  3. Servers and their operation
    1. Registration
    2.  SIP server in Proxy and Redirect modes
    3. Stateless and stateful Proxy servers
    4. Location server
    5. SRV records and DNS
    6. uri/url/urn, ENUM and NAPTR records
  4. SIP signalling messages (including Instant Messaging & Presence – IMP extensions)
    1. Message structure
    2. Requests
    3. Responses
    4. Example of a call
    5. Headers and parameters
    6. IMP models
  5. SDP (Session Description Protocol)
    1. Description of media
    2. Standard list of 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
    4. Changing media during a session
    5. Using 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 signalling
  9. SIP – security problems
    1. Secure SIP, Secure RTP and Secure RTCP
    2. Typical implementations of Secure SIP
  10. Practical problems and perspectives
    1. NAT and firewall traversal
    2. QoS
    3. SIP and SDP in 3GPP IMS architecture
  11. Wrap-up and discussion

Part II: Hands on

  1. SIP in LAN environment: XLite SIP UA + Asterisk
    1. Creating Asterisk accounts with a simple 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. P2P calls with Linphone
  2. Analyzing of SIP signalling using Wireshark
    1. Configuration of a server
    2. Registration of SIP signalling and RTP media streams
    3. SIP packet analysis. Retrieval of a specific call
    4. Voice quality problems. Jitter buffer. Retrieval of DTMF signalling (RFC 2833, INFO). Codec and DTMF troubleshooting (transcoding, GSM codec failure, DTMF tone duplication)
    5. VoIP monitor
  3. SDP, Instant Messaging and Presence (IM&P)
    1. SDP parameters and attributes
    3. Practising IM&P 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
      • Analysing SIP call signalling using Wireshark
      • Troubleshooting – DNS timeout, 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 – 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
      • 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 dependent constraints
  9. SIP & Network Adress 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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