Szkolenia Real Time Services over IP

Szkolenia Real Time Services over IP

Real Time Services over IP courses

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Practical Voice and Video over IP

b.d

Ireneusz Krawczyk - Gigaset Communications Polska Sp. z o.o.

Plany Szkoleń Real Time Services over IP

Kod Nazwa Czas trwania Charakterystyka kursu
vvoip Practical Voice and Video over IP 21 godz. A 3 day practical course covering the fundamentals of the various voice protocols used to carry voice (and video) over a packet switched network. This course is a mixture of theory and practice (utilising Wireshark where appropriate for explanation and troubleshooting) with practical VoIP configured using IP telephones, softphones and voice capable Cisco routers. This is primarily a ‘generic’ VoIP course – but uses Cisco equipment to provide an IP backbone and voice gateways. We can also provide Cisco specific voice training courses. The course also covers call setup between circuit switched and packet switched networks. The course is approximately 40% practical. Objectives: Explain the operation of circuit switched networks. Contrast the differences between circuit switched and packet switched networks. Explain the fields and flags within the IP header – including the new DSCP field. Explain and quantify the need for voice compression and delay in an IP network. Identify the CODEC’s used in voice and video compression – e.g. G711, G729 and H261/3/4. Identify and explain the fields of the Real Time Transport Protocol and Real Time Control Protocol – including carrying DTMF tones. Calculate bandwidth requirements for various voice compression algorithms. Explain and demonstrate the need for security and encryption within VoIP. Describe and explain the H.323 protocol stack. Describe and explain call setup and signalling (H.225). Describe and explain H.245, fast start and tunnelling of H245. Describe and explain SIP call setup. Describe and explain SIP Servers. Describe and explain SDP. Describe and explain Soft Switching – including MGCP and H248. Describe and explain the use of SCTP in backhauling signalling messages. Describe and explain the importance of network delay and the need for QoS in the network. Describe and explain the differences between reservation and prioritisation QoS in an IP Network. Describe the use of IntServ and RSVP in QoS. Describe the use of TOS and DiffServ in QoS. Describe and explain the need for multiple queues and intelligent scheduling of packets at router/switch interfaces. Describe the use of MPLS to deliver QoS in the WAN environment. Describe the use of VLAN’s to deliver QoS in the LAN environment. Practical Exercises: Lab Exercise 1: Practical VoIP in the LAN. Lab Exercise 2: Build IP network, make calls and examine RTP packets using Wireshark. Lab Exercise 3: Make calls and examine SIP Packets using Wireshark. Lab Exercise 4: Make calls and examine H323 Packets using Wireshark. Lab Exercise 5: Calls between PSTN and IP Networks. Lab Exercise 6: Configure QoS and make calls to examine effects. Lab Exercise 7: Security and VoIP. Equipment used in Practical Sessions: Cisco 2600 series and 2800 series voice capable routers and Cisco ASA 5505 Firewalls. IP Telephones, softphones and analogue telephones as required. SIP client and server software as required.
siptel Practical SIP Telephony 21 godz. A 3 day practical course covering the messages and call flows of the Session Initiation Protocol (SIP) and its use in voice networks. This course is a mixture of theory and practice (utilising protocol analyser traces where appropriate for explanation and troubleshooting) with practical VoIP configured using IP telephones, softphones, voice capable Cisco routers and SIP IP PBX,s (e.g. Trixbox). Audience: Network Planners, Designers, and Engineers requiring an understanding of SIP. The course is around 40% practical. Objectives: Describe call signalling and setup in the voice network Describe carrying of voice media and bandwidth requirements for VoIP calls Describe SIP standards, services, messages and return codes Describe basic call setup using SIP Describe SIP flows and SDP Describe registration process and making calls with a SIP Server Describe IP PBX and Call Conferences Describe SRV records and DNS Describe uri/url/urn, ENUM and NAPTR Records Describe mapping of services to an address Describe SIP-T and SIP-I Describe SIP early media and SIP trunks Describe call flows between PSTN and IP using SIP Describe Secure SIP, Secure RTP and Secure RTCP Describe typical Secure SIP implementations Practical Exercises: Lab Exercise 1: Practical SIP in the LAN with Xlite Lab Exercise 2: Examine SIP Packets using Wireshark Lab Exercise 3: SDP, Presence and IM Lab Exercise 4: Call Flows with SIP Server Lab Exercise 5a: SIP Registration with DNS Lab Exercise 5b: Call Flows with DNS Lab Exercise 6: SIP Trunks Lab Exercise 7: Security with IPSec Lab Exercise 8: Security with Secure SIP

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