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Eğitim İçeriği
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 (trancoding, 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 Address 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
21 Saat
Danışanlarımızın Yorumları (7)
Goegzersiz uzmanı ile iletişime geçin
Radziszewski - EduBroker Sp. z o.o.
Eğitim - SIP protocol in VoIP
Yapay Zeka Çevirisi
Ayrıntılı提供的痕迹分析被翻译成了目标语言，但由于指导方针要求保持原文结构和格式且此处的中文是中间过程的误植，正确的翻译应直接作用于给定的英文短语： Provided traces'in detaylı analizi 请注意，根据指南，我将仅翻译所提供的内容，并保持其原始结构。因此，正确的响应应该是直接翻译给定的英语短语，而不包含任何额外的解释或格式更改。
Krzysztof - EduBroker Sp. z o.o.
Eğitim - SIP protocol in VoIP
Yapay Zeka Çevirisi
örneklere yer verildi
Slawomir - NBP
Eğitim - SIP protocol in VoIP
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Instructor her laboratuvar oturumunda adımları açıklamak ve göstermek için sabırlıdır.
CHEE WAI KAN - Maxis Broadband Sdn Bhd
Eğitim - SIP protocol in VoIP
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Eğitmen sorulan sorular için açıklama yapabilir ve örnek verebilir.
SHAMSANI ABU BAKAR - Maxis Broadband Sdn Bhd
Eğitim - SIP protocol in VoIP
Yapay Zeka Çevirisi
Çok bilgilendirici ve detaylı açıklama
Syahrifendi Lukmanilhakim - Maxis Broadband Sdn Bhd
Eğitim - SIP protocol in VoIP
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Slaytın araçları ve içeriği.
Kak Khen Man - Maxis Broadband Sdn Bhd
Eğitim - SIP protocol in VoIP
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