Thank you for sending your enquiry! One of our team members will contact you shortly.
Thank you for sending your booking! One of our team members will contact you shortly.
Course Outline
Part I: Introduction
- Introduction
- History and motivation
- VoIP types and evolution
- SIP core concepts
- SIP standardization (RFC 3261 and related standards)
- Architecture
- User Agent (UA)
- Standard servers: Registrar, Location, Proxy, and Redirect
- Application servers
- Identification and addressing
- SIP topology
- Servers and operations
- Registration process
- SIP server in Proxy and Redirect modes
- Stateless vs. stateful Proxy servers
- Location server functionality
- SRV records and DNS
- URI/URL/URN, ENUM, and NAPTR records
- SIP signaling messages (including Instant Messaging & Presence - IMP extensions)
- Message structure
- Requests
- Responses
- Call example
- Headers and parameters
- IMP models
- SDP (Session Description Protocol)
- Media description
- Standard codec list
- Session negotiation rules
- Call flows - SIP signaling
- SIP session - main RFC 3261 example
- Sample call scenarios
- Conferencing and IP PBX
- Media modification during a session
- Using IMP
- SIP request and response routing
- VIA header
- ROUTE and RECORD-ROUTE headers
- SIP-PSTN interworking
- SIP-T and SIP-I
- SIP early media and trunking
- SIP-PSTN signaling
- SIP security challenges
- Secure SIP, Secure RTP, and Secure RTCP
- Common implementations of Secure SIP
- Practical issues and future perspectives
- NAT and firewall traversal
- Quality of Service (QoS)
- SIP and SDP in 3GPP IMS architecture
- Wrap-up and discussion
Part II: Hands-on Lab
- SIP in LAN environment: XLite SIP UA + Asterisk
- Creating Asterisk accounts with a basic dial plan
- Configuring XLite SIP UA (DTMF, codecs, NAT, RTP, timers, registration) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
- Registration, initiating, and receiving calls
- P2P calls using Linphone
- Analyzing SIP signaling with Wireshark
- Server configuration
- Capturing SIP signaling and RTP media streams
- SIP packet analysis. Retrieving specific call details
- Voice quality issues. Jitter buffer. Retrieving DTMF signaling (RFC 2833, INFO). Codec and DTMF troubleshooting (transcoding, GSM codec failure, DTMF tone duplication)
- VoIP monitoring
- SDP, Instant Messaging, and Presence (IM&P)
- SDP parameters and attributes
- SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
- Practicing IM&P with XLite and Linphone
- SIP call flows
- SIP Registration with DNS
- SIP SRV record
- SIP phone registration using DNS-SRV
- Call Flows with DNS
- Analyzing SIP call signaling using Wireshark
- Troubleshooting - DNS timeout, latency
- SIP Registration with DNS
- SIP trunks
- Establishing a test SIP trunk
- Troubleshooting (DoS, DDoS, fraud, calls per second)
- SIP security issues
- SIP security with IPSec
- Security with Secure SIP
- IP telephony - fraud risks
- Preventing DDoS and other attacks
- Deploying SIP-based VoIP services
- Switch configuration
- SIP client configuration and registration
- Software
- Asterisk PBX / FreeSWITCH softswitch / Cisco Call Manager
- Linux CentOS
- TDM2IP drivers
- Softphones (XLite, Linphone)
- Hardware
- Server
- TDM2IP card/gateway
- Hardphone (Polycom, Gigaset, Yealink)
- Softphone/Hardphone
- Configuration
- Codecs
- User/Password/SIP Server/Proxy/Ports
- Operation and signaling for:
- 3-Way Calling
- Call Forwarding
- Attendant Call Transfer
- MWI, BLF
- Yealink auto-provisioning
- Vendor-specific constraints
- Configuration
- SIP & Network Address Translation (NAT) challenges
- NAT types and structure
- STUN (Simple Traversal of UDP Through NATs)
- VoIP call quality - troubleshooting
- Call connected but missing media
- Key QoS factors
- Delay, jitter, play buffer size
- VoIP quality metrics
- RTCP - delay and jitter
- MOS according to ITU-T G.107 E-model
- VoIP quality monitoring tools (Voipmonitor)
- Cloud-based IP telephony
- Wrap-up and addressing SIP and VoIP issues submitted by participants
21 Hours
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.