Course Outline
Part I: Introduction
- Introduction
- History and motivation
- Types of VoIP and its evolution
- SIP – main concepts
- SIP standardization (RFC 3261 and other relevant standards)
- Architecture
- UA – User Agent
- Predefined servers: Registrar, Location, Proxy and Redirect
- Application servers
- Identification and addressing
- SIP trapezoid
- Servers and their operation
- Registration
- SIP server in Proxy and Redirect modes
- Stateless and stateful Proxy servers
- Location server
- SRV records and DNS
- uri/url/urn, ENUM and NAPTR records
- SIP signalling messages (including Instant Messaging & Presence – IMP extensions)
- Message structure
- Requests
- Responses
- Example of a call
- Headers and parameters
- IMP models
- SDP (Session Description Protocol)
- Description of media
- Standard list of codecs
- Session negotiation rules
- Call flows – SIP signalling
- SIP session – main RFC 3261 example
- Sample call scenarios
- Conferencing and IP PBX
- Changing media during a session
- Using IMP
- Routing of SIP requests and responses
- VIA header
- ROUTE and RECORD-ROUTE headers
- SIP-PSTN interworking
- SIP-T and SIP-I
- SIP early media and SIP trunking
- SIP-PSTN signalling
- SIP – security problems
- Secure SIP, Secure RTP and Secure RTCP
- Typical implementations of Secure SIP
- Practical problems and perspectives
- NAT and firewall traversal
- QoS
- SIP and SDP in 3GPP IMS architecture
- Wrap-up and discussion
Part II: Hands on
- SIP in LAN environment: XLite SIP UA + Asterisk
- Creating Asterisk accounts with a simple dial plan
- Configuration of XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
- Registration, initiating and receiving calls
- P2P calls with Linphone
- Analyzing of SIP signalling using Wireshark
- Configuration of a server
- Registration of SIP signalling and RTP media streams
- SIP packet analysis. Retrieval of a specific call
- Voice quality problems. Jitter buffer. Retrieval of DTMF signalling (RFC 2833, INFO). Codec and DTMF troubleshooting (transcoding, GSM codec failure, DTMF tone duplication)
- VoIP monitor
- SDP, Instant Messaging and Presence (IM&P)
- SDP parameters and attributes
- SUBSCRIBE, PUBLISH and MESSAGE SIP methods
- Practising 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
- Analysing SIP call signalling using Wireshark
- Troubleshooting – DNS timeout, latency
- SIP Registration with DNS
- SIP trunks
- Establishing a test SIP trunk
- Troubleshooting (DOS, DDOS, fraud, cps)
- SIP security issues
- SIP security with IPSec
- Security with Secure SIP
- IP telephony – risk of frauds
- Preventing DDOS and other types of attacks
- Launching SIP based VoIP services
- Configuration of a switch
- 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 signalling for:
- 3-Way Calling
- Call Forwarding
- Attendant Call Transfer
- MWI, BLF
- Yealink autoprovisioning
- Vendor dependent constraints
- Configuration
- SIP & Network Adress Translation (NAT) problems
- Type and structure of NATs
- STUN (Simple Traversal of UDP Through NATs)
- Quality of VoIP calls – troubleshooting
- Call connected – 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 related issues submitted by participants
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
topics related to NAT, STUN, security, VOIP solutions
Dzmitry - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
Machine Translated
The knowledge of the presenter was impressive. We received very comprehensive answers to all our questions and doubts.
Olgierd Januszkiewicz - NBP
Course - SIP protocol in VoIP
Machine Translated
examples presented
Slawomir - NBP
Course - SIP protocol in VoIP
Ease and ease of conducting training.
Jaroslaw - Politechnika Wroclawska
Course - SIP protocol in VoIP
Machine Translated
the practical part, which was definitely too little, and the instructor's help in carrying out current tasks.
Robert - Polska Spolka Gazownictwa sp. z o.o.
Machine Translated