-
Voice communication over IP is going to be the technology of the future. No matter whether it is in the corporate environment or for providers—soon all services relating to voice and video will be placed on an IP platform. If voice and IP merge, specialists from telecommunications and network sectors have to work our solutions together. This course offers a detailed introduction into Voice over IP (VoIP) for both sides. It includes the concepts of voice communication over IP and deals with RTP as the most important protocol for voice transmission as well as with the two signaling protocols SIP and H.323. Further focal points are Quality of Service and solutions for fax transmission. The course introduces into the planning and design of VoIP solutions for companies of differing sizes and shows migration strategies for future PSTN connections over SIP trunks. One day with hands-on exercises serves to explain the functioning of VoIP solutions.
-
Course Contents
-
- Voice over IP—Basics, Concepts, and Protocols
- Basics of Voice Communication
- Codecs and Bandwidths for VoIP/IP Telephony
- Media Streams over IP-RTP
- Basics of SIP—Terms, Concepts, and Processes
- Signaling over SIP—Registration and Call Setup
- Negotiation of Media Streams over SDP
- Basics of H.323—H.323 Gatekeeper and H.323 Terminals
- Signaling over H.323—RAS, H.225, and H.245
- Media Gateways over MGCP and H.248/Megaco
- VoIP in Practical Application—Quality of Service and Fax Transmission over IP
- Basics of VoIP Security—Encryption, Firewalls, and NAT
- VoIP Design—Concepts for Small, Medium-Sized and Large Companies
- Cloud, Hosting, or IP-Centrex—The PBX at the Provider
- SIP Trunking—The VoIP Connection to the Provider
- Future Trends in Telephony
One day of hands-on exercises and the analysis of traces will make the course contents easier to understand.
You will receive the comprehensive documentation package of the ExperTeach Networking series – printed documentation, e-book, and personalized PDF! As online participant, you will receive the e-book and the personalized PDF.
-
Target Group
-
The course addresses designers, consultants, decision-takers, and technicians from the areas of telecommunications and network technology who are looking for a basic introduction into the world of Voice over IP. It offers solid information required to plan and implement the migration to VoIP.
-
Knowledge Prerequisites
-
Basic knowledge of the telecommunications and IP world is mandatory for attendance at this course.
-
Complementary and Continuative Courses
-
Course Objective
-
After the course, the students will be able to assess VoIP concepts, plan and implement migration concepts, and deepen their product-specific knowledge.
| 1 | Introduction and Motivation |
| 1.1 | Voice Networks of Today and Tomorrow |
| 1.1.1 | User Trends |
| 1.1.2 | Trends in the Enterprise Market |
| 1.1.3 | Trends among Providers |
| 1.1.4 | Trends in the Data Centers |
| 1.1.5 | All IP—Internet for Everything |
| 1.2 | Voice over IP—Basics, Concepts, and Protocols |
| 1.2.1 | VoIP Protocols |
| 1.2.2 | VoIP in the ISO/OSI Model |
| 1.2.3 | VoIP Signaling |
| 1.2.4 | Media Streams |
| 1.3 | VoIP Infrastructure and Application Areas |
| 1.3.1 | VoIP in the Enterprise Environment |
| 1.3.2 | VoIP for Private Customers |
| 1.3.3 | VoIP in the Provider Environment |
| 1.3.4 | VoIP over the Internet |
| 1.3.5 | WebRTC |
| 2 | Media Streams with RTP |
| 2.1 | Voice Transfer |
| 2.1.1 | Digitizing Voice |
| 2.1.2 | Codecs—PCM and More |
| 2.1.3 | Hybrid Encoding via CELP and MP-MLQ |
| 2.2 | Transporting Voice over IP |
| 2.2.1 | The Anatomy of RTP Packets |
| 2.2.2 | IP Addressing and Routing |
| 2.2.3 | The Transport Protocols |
| 2.3 | Real-time Transport Protocol (RTP) |
| 2.3.1 | Demands Made on RTP |
| 2.3.2 | The Frame Format of RTP |
| 2.3.3 | RTP Profiles |
| 2.4 | Real-Time Transport Control Protocol (RTCP) |
| 2.4.1 | Classic RTCP |
| 2.4.2 | RTCP Extended Reports (RTCP XR) |
| 2.5 | RTP Applications |
| 2.5.1 | Dial Tones via DTMF |
| 2.5.2 | Speech Pauses and VAD |
| 2.5.3 | Header Compression with cRTP |
| 2.5.4 | Bandwidths for VoIP |
| 2.6 | Parameters Influencing Voice Quality |
| 2.6.1 | Delay—End-to-End |
| 2.6.2 | Jitter and Jitter Buffer |
| 2.6.3 | Packet Loss and Packet Loss Concealment |
| 2.7 | Voice Quality—Models and Calculation |
| 2.7.1 | Mean Opinion Score (MOS) |
| 2.7.2 | Objective, but Old: PAMS |
| 2.7.3 | Objective Perception: PSQM and PESQ |
| 2.7.4 | Subjective Perception: E-Model with R Factor |
| 2.7.5 | POLQA and TOSQA |
| 3 | SIP—The Session Initiation Protocol |
| 3.1 | SIP—An Overview |
| 3.1.1 | Standardization |
| 3.1.2 | SIP in the ISO/OSI Model |
| 3.1.3 | SIP Addressing: SIP URI and TEL URI |
| 3.2 | The Components of the SIP Architecture |
| 3.2.1 | The End Devices: User Agents |
| 3.2.2 | The SIP Proxy |
| 3.2.3 | SIP Gateways |
| 3.3 | Protocol Setup |
| 3.3.1 | Setup of SIP Messages |
| 3.3.2 | SIP Requests—SIP Methods |
| 3.3.3 | SIP Responses |
| 3.3.4 | The Message Body |
| 3.4 | SDP—The Session Description Protocol |
| 3.5 | Registration and Authentication |
| 3.5.1 | SIP Registration—Processes |
| 3.5.2 | SIP-Register without Authentication |
| 3.5.3 | Register with Authentication |
| 3.6 | SIP Call Setup with Proxy |
| 3.6.1 | SIP Invite over the Classic Proxy |
| 3.6.2 | SIP Server Terminates the Dialog |
| 3.6.3 | Domain Environments and DNS |
| 3.7 | Deployment of SIP Today and Tomorrow |
| 4 | Gateway Concepts for VoIP |
| 4.1 | Gateway Control |
| 4.2 | H.323 Used in Companies |
| 4.2.1 | H.323 Implementation |
| 4.2.2 | H.323 Architecture |
| 4.2.3 | Call Setup from SIP to H.323 |
| 4.3 | MGCP |
| 4.3.1 | Application Scenario Enterprise |
| 4.3.2 | Application Scenario Provider |
| 4.3.3 | MGCP—The Protocol Setup |
| 4.3.4 | Call Setup over MGCP |
| 4.3.5 | Call Setup—MGCP to SIP |
| 4.4 | H.248/Megaco |
| 4.4.1 | Termination and Context |
| 4.4.2 | Commands |
| 4.4.3 | Descriptors |
| 5 | VoIP—Practical Application |
| 5.1 | Encryption |
| 5.1.1 | Encryption of the Signaling over SIPS |
| 5.1.2 | Encryption of the Media Stream via SRTP |
| 5.1.3 | Key Management in the Session Description Protocol |
| 5.1.4 | Encryption between Sites |
| 5.2 | VoIP with NAT and Firewalls |
| 5.2.1 | VoIP and Stateful Firewalls |
| 5.2.2 | VoIP and NAT |
| 5.2.3 | Solution #1: Application Layer Gateway (ALG) |
| 5.2.4 | Solution #2: STUN, TURN, and ICE |
| 5.2.5 | Solution #3: Hosted NAT Traversal (HNT) |
| 5.2.6 | Solution #4: Enterprise SBC |
| 5.3 | Fax Transmission over IP |
| 5.3.1 | Special Features during Fax Transmission |
| 5.3.2 | Fax Transmission Process |
| 5.3.3 | Fax as a Default VoIP Call |
| 5.3.4 | T.37—Fax as E-Mail Attachment |
| 5.3.5 | T.38—Fax in Real-Time |
| 5.3.6 | Error Patterns with Fax over IP |
| 5.4 | Quality of Service |
| 5.4.1 | What is Quality of Service? |
| 5.4.2 | Classification and Tagging |
| 5.4.3 | Queuing |
| 5.4.4 | Policing |
| 5.4.5 | Traffic Shaping |
| 5.4.6 | Admission Control |
| 6 | Concepts and Application Scenarios on the Enterprise Sector |
| 6.1 | Questions and Concepts |
| 6.1.1 | Access to Telephony Service Providers |
| 6.1.2 | Emergency Calls |
| 6.1.3 | Features |
| 6.1.4 | Features for VoIP vs. PSTN |
| 6.2 | Enterprise Solutions for a Site |
| 6.2.1 | Voice VLANs and PoE |
| 6.3 | Enterprise Solutions for Several Sites |
| 6.3.1 | WAN Interconnection—Private or Public |
| 6.3.2 | Central PBX |
| 6.3.3 | Decentralized PBXs |
| 6.4 | IP Centrex—Cloud PBX or Hosted PBX |
| 6.5 | Access to Telephony Service Providers over SIP Trunks |
| 6.5.1 | SIP Trunking Concept |
| 6.5.2 | Integration of the SBC—Stand-alone Devices |
| 6.5.3 | Registration Mode and Static Mode |
| 6.5.4 | Registration at the SIP Trunk |
| 6.5.5 | Identities: P-Asserted Identity and From: |
| 6.5.6 | Signaling at the SIP Trunk |
| 6.6 | Migration |
| 6.6.1 | Approach—Smooth Migration |
| 6.6.2 | Approach—Hard Migration |
| 6.6.3 | Approach—Site-wise Migration |
| 6.6.4 | Migration Scenario |
| A | Lab Exercises |
| A.1 | Lab Setup for Practical Exercises |
| A.1.1 | Lab Setup for Demonstration Purposes |
| A.1.2 | Lab Setup with Softphones |
| A.1.3 | Hardware and Software |
| A.2 | Hands-on Exercises |
| A.2.1 | Lab Exercise—Registration |
| A.2.2 | Lab Exercise—Basic Call with SIP |
| A.2.3 | Lab Exercise—Call Hold |
| A.2.4 | Lab Exercise—Call Transfer |
| A.2.5 | Lab Exercise—Simple Three-Party Conference |
| A.2.6 | Lab Exercise—Videotelephony |
| A.2.7 | Lab Exercise—Selection of the Codec |
| A.2.8 | Lab Exercise—Incompatible Codecs |
| A.3 | Wireshark in a Brief Overview |
| A.3.1 | Capturing with Wireshark |
| A.3.2 | Preferences |
| A.3.3 | Preferences and Profiles |
| A.3.4 | Display Filters |
| A.4 | Evaluation of RTP with Wireshark |
| A.4.1 | RTP Statistics |
| A.4.2 | RTP Stream Analysis |
| A.4.3 | Lab Exercise: RTP Basic Functions |
| A.4.4 | Lab Exercise: RTP Operation |
| A.5 | SIP Analysis with Wireshark |
| A.5.1 | VoIP Calls—Statistics |
| A.5.2 | SIP Statistics |
| A.5.3 | Lab Exercise: SIP Registration |
| A.5.4 | Lab Exercise: SIP—Basic Call with Wireshark |
| B | H.323—Terms and Procedures |
| B.1 | H.323 Architecture |
| B.1.1 | H.323 Procedures in the TCP/IP Protocol Stack |
| B.1.2 | H.323 Terminal—The Functions of the End Devices |
| B.1.3 | H.323 Gatekeeper—Address Translation and Management |
| B.1.4 | H.323 Gateway—The Translator |
| B.1.5 | The H.323 MCU—Conference Calls |
| B.2 | RAS—Services |
| B.2.1 | H.323 Identities and Addressing |
| B.2.2 | Registration with H.323 |
| B.2.3 | RAS Channel: Admission |
| B.3 | Call Setup and Termination |
| B.3.1 | H.225—Call Signaling Channel with Setup |
| B.3.2 | H.245 |
| B.3.3 | Call Termination |
| B.3.4 | A Complete Call |
| B.3.5 | H.323 Options: Fast Connect and Tunneling |
| B.4 | H.235—Encryption for H.323 |
| B.5 | H.323 and Firewalls |
| C | List of Abbreviations |
-
Classroom training
- Do you prefer the classic training method? A course in one of our Training Centers, with a competent trainer and the direct exchange between all course participants? Then you should book one of our classroom training dates!
-
Hybrid training
- Hybrid training means that online participants can additionally attend a classroom course. The dynamics of a real seminar are maintained, and the online participants are able to benefit from that. Online participants of a hybrid course use a collaboration platform, such as WebEx Training Center or Saba Meeting. To do this, a PC with browser and Internet access is required, as well as a headset and ideally a Web cam. In the seminar room, we use specially developed and customized audio- and video-technologies. This makes sure that the communication between all persons involved works in a convenient and fault-free way.
-
Online training
- You wish to attend a course in online mode? We offer you online course dates for this course topic. To attend these seminars, you need to have a PC with Internet access (minimum data rate 1Mbps), a headset when working via VoIP and optionally a camera. For further information and technical recommendations, please refer to.
-
Tailor-made courses
-
You need a special course for your team? In addition to our standard offer, we will also support you in creating your customized courses, which precisely meet your individual demands. We will be glad to consult you and create an individual offer for you.
Request for customized courses
You can find the complete description of this course with dates and prices ready for download at as PDF.-
Voice communication over IP is going to be the technology of the future. No matter whether it is in the corporate environment or for providers—soon all services relating to voice and video will be placed on an IP platform. If voice and IP merge, specialists from telecommunications and network sectors have to work our solutions together. This course offers a detailed introduction into Voice over IP (VoIP) for both sides. It includes the concepts of voice communication over IP and deals with RTP as the most important protocol for voice transmission as well as with the two signaling protocols SIP and H.323. Further focal points are Quality of Service and solutions for fax transmission. The course introduces into the planning and design of VoIP solutions for companies of differing sizes and shows migration strategies for future PSTN connections over SIP trunks. One day with hands-on exercises serves to explain the functioning of VoIP solutions.
-
Course Contents
-
- Voice over IP—Basics, Concepts, and Protocols
- Basics of Voice Communication
- Codecs and Bandwidths for VoIP/IP Telephony
- Media Streams over IP-RTP
- Basics of SIP—Terms, Concepts, and Processes
- Signaling over SIP—Registration and Call Setup
- Negotiation of Media Streams over SDP
- Basics of H.323—H.323 Gatekeeper and H.323 Terminals
- Signaling over H.323—RAS, H.225, and H.245
- Media Gateways over MGCP and H.248/Megaco
- VoIP in Practical Application—Quality of Service and Fax Transmission over IP
- Basics of VoIP Security—Encryption, Firewalls, and NAT
- VoIP Design—Concepts for Small, Medium-Sized and Large Companies
- Cloud, Hosting, or IP-Centrex—The PBX at the Provider
- SIP Trunking—The VoIP Connection to the Provider
- Future Trends in Telephony
One day of hands-on exercises and the analysis of traces will make the course contents easier to understand.
You will receive the comprehensive documentation package of the ExperTeach Networking series – printed documentation, e-book, and personalized PDF! As online participant, you will receive the e-book and the personalized PDF.
-
Target Group
-
The course addresses designers, consultants, decision-takers, and technicians from the areas of telecommunications and network technology who are looking for a basic introduction into the world of Voice over IP. It offers solid information required to plan and implement the migration to VoIP.
-
Knowledge Prerequisites
-
Basic knowledge of the telecommunications and IP world is mandatory for attendance at this course.
-
Complementary and Continuative Courses
-
Course Objective
-
After the course, the students will be able to assess VoIP concepts, plan and implement migration concepts, and deepen their product-specific knowledge.
| 1 | Introduction and Motivation |
| 1.1 | Voice Networks of Today and Tomorrow |
| 1.1.1 | User Trends |
| 1.1.2 | Trends in the Enterprise Market |
| 1.1.3 | Trends among Providers |
| 1.1.4 | Trends in the Data Centers |
| 1.1.5 | All IP—Internet for Everything |
| 1.2 | Voice over IP—Basics, Concepts, and Protocols |
| 1.2.1 | VoIP Protocols |
| 1.2.2 | VoIP in the ISO/OSI Model |
| 1.2.3 | VoIP Signaling |
| 1.2.4 | Media Streams |
| 1.3 | VoIP Infrastructure and Application Areas |
| 1.3.1 | VoIP in the Enterprise Environment |
| 1.3.2 | VoIP for Private Customers |
| 1.3.3 | VoIP in the Provider Environment |
| 1.3.4 | VoIP over the Internet |
| 1.3.5 | WebRTC |
| 2 | Media Streams with RTP |
| 2.1 | Voice Transfer |
| 2.1.1 | Digitizing Voice |
| 2.1.2 | Codecs—PCM and More |
| 2.1.3 | Hybrid Encoding via CELP and MP-MLQ |
| 2.2 | Transporting Voice over IP |
| 2.2.1 | The Anatomy of RTP Packets |
| 2.2.2 | IP Addressing and Routing |
| 2.2.3 | The Transport Protocols |
| 2.3 | Real-time Transport Protocol (RTP) |
| 2.3.1 | Demands Made on RTP |
| 2.3.2 | The Frame Format of RTP |
| 2.3.3 | RTP Profiles |
| 2.4 | Real-Time Transport Control Protocol (RTCP) |
| 2.4.1 | Classic RTCP |
| 2.4.2 | RTCP Extended Reports (RTCP XR) |
| 2.5 | RTP Applications |
| 2.5.1 | Dial Tones via DTMF |
| 2.5.2 | Speech Pauses and VAD |
| 2.5.3 | Header Compression with cRTP |
| 2.5.4 | Bandwidths for VoIP |
| 2.6 | Parameters Influencing Voice Quality |
| 2.6.1 | Delay—End-to-End |
| 2.6.2 | Jitter and Jitter Buffer |
| 2.6.3 | Packet Loss and Packet Loss Concealment |
| 2.7 | Voice Quality—Models and Calculation |
| 2.7.1 | Mean Opinion Score (MOS) |
| 2.7.2 | Objective, but Old: PAMS |
| 2.7.3 | Objective Perception: PSQM and PESQ |
| 2.7.4 | Subjective Perception: E-Model with R Factor |
| 2.7.5 | POLQA and TOSQA |
| 3 | SIP—The Session Initiation Protocol |
| 3.1 | SIP—An Overview |
| 3.1.1 | Standardization |
| 3.1.2 | SIP in the ISO/OSI Model |
| 3.1.3 | SIP Addressing: SIP URI and TEL URI |
| 3.2 | The Components of the SIP Architecture |
| 3.2.1 | The End Devices: User Agents |
| 3.2.2 | The SIP Proxy |
| 3.2.3 | SIP Gateways |
| 3.3 | Protocol Setup |
| 3.3.1 | Setup of SIP Messages |
| 3.3.2 | SIP Requests—SIP Methods |
| 3.3.3 | SIP Responses |
| 3.3.4 | The Message Body |
| 3.4 | SDP—The Session Description Protocol |
| 3.5 | Registration and Authentication |
| 3.5.1 | SIP Registration—Processes |
| 3.5.2 | SIP-Register without Authentication |
| 3.5.3 | Register with Authentication |
| 3.6 | SIP Call Setup with Proxy |
| 3.6.1 | SIP Invite over the Classic Proxy |
| 3.6.2 | SIP Server Terminates the Dialog |
| 3.6.3 | Domain Environments and DNS |
| 3.7 | Deployment of SIP Today and Tomorrow |
| 4 | Gateway Concepts for VoIP |
| 4.1 | Gateway Control |
| 4.2 | H.323 Used in Companies |
| 4.2.1 | H.323 Implementation |
| 4.2.2 | H.323 Architecture |
| 4.2.3 | Call Setup from SIP to H.323 |
| 4.3 | MGCP |
| 4.3.1 | Application Scenario Enterprise |
| 4.3.2 | Application Scenario Provider |
| 4.3.3 | MGCP—The Protocol Setup |
| 4.3.4 | Call Setup over MGCP |
| 4.3.5 | Call Setup—MGCP to SIP |
| 4.4 | H.248/Megaco |
| 4.4.1 | Termination and Context |
| 4.4.2 | Commands |
| 4.4.3 | Descriptors |
| 5 | VoIP—Practical Application |
| 5.1 | Encryption |
| 5.1.1 | Encryption of the Signaling over SIPS |
| 5.1.2 | Encryption of the Media Stream via SRTP |
| 5.1.3 | Key Management in the Session Description Protocol |
| 5.1.4 | Encryption between Sites |
| 5.2 | VoIP with NAT and Firewalls |
| 5.2.1 | VoIP and Stateful Firewalls |
| 5.2.2 | VoIP and NAT |
| 5.2.3 | Solution #1: Application Layer Gateway (ALG) |
| 5.2.4 | Solution #2: STUN, TURN, and ICE |
| 5.2.5 | Solution #3: Hosted NAT Traversal (HNT) |
| 5.2.6 | Solution #4: Enterprise SBC |
| 5.3 | Fax Transmission over IP |
| 5.3.1 | Special Features during Fax Transmission |
| 5.3.2 | Fax Transmission Process |
| 5.3.3 | Fax as a Default VoIP Call |
| 5.3.4 | T.37—Fax as E-Mail Attachment |
| 5.3.5 | T.38—Fax in Real-Time |
| 5.3.6 | Error Patterns with Fax over IP |
| 5.4 | Quality of Service |
| 5.4.1 | What is Quality of Service? |
| 5.4.2 | Classification and Tagging |
| 5.4.3 | Queuing |
| 5.4.4 | Policing |
| 5.4.5 | Traffic Shaping |
| 5.4.6 | Admission Control |
| 6 | Concepts and Application Scenarios on the Enterprise Sector |
| 6.1 | Questions and Concepts |
| 6.1.1 | Access to Telephony Service Providers |
| 6.1.2 | Emergency Calls |
| 6.1.3 | Features |
| 6.1.4 | Features for VoIP vs. PSTN |
| 6.2 | Enterprise Solutions for a Site |
| 6.2.1 | Voice VLANs and PoE |
| 6.3 | Enterprise Solutions for Several Sites |
| 6.3.1 | WAN Interconnection—Private or Public |
| 6.3.2 | Central PBX |
| 6.3.3 | Decentralized PBXs |
| 6.4 | IP Centrex—Cloud PBX or Hosted PBX |
| 6.5 | Access to Telephony Service Providers over SIP Trunks |
| 6.5.1 | SIP Trunking Concept |
| 6.5.2 | Integration of the SBC—Stand-alone Devices |
| 6.5.3 | Registration Mode and Static Mode |
| 6.5.4 | Registration at the SIP Trunk |
| 6.5.5 | Identities: P-Asserted Identity and From: |
| 6.5.6 | Signaling at the SIP Trunk |
| 6.6 | Migration |
| 6.6.1 | Approach—Smooth Migration |
| 6.6.2 | Approach—Hard Migration |
| 6.6.3 | Approach—Site-wise Migration |
| 6.6.4 | Migration Scenario |
| A | Lab Exercises |
| A.1 | Lab Setup for Practical Exercises |
| A.1.1 | Lab Setup for Demonstration Purposes |
| A.1.2 | Lab Setup with Softphones |
| A.1.3 | Hardware and Software |
| A.2 | Hands-on Exercises |
| A.2.1 | Lab Exercise—Registration |
| A.2.2 | Lab Exercise—Basic Call with SIP |
| A.2.3 | Lab Exercise—Call Hold |
| A.2.4 | Lab Exercise—Call Transfer |
| A.2.5 | Lab Exercise—Simple Three-Party Conference |
| A.2.6 | Lab Exercise—Videotelephony |
| A.2.7 | Lab Exercise—Selection of the Codec |
| A.2.8 | Lab Exercise—Incompatible Codecs |
| A.3 | Wireshark in a Brief Overview |
| A.3.1 | Capturing with Wireshark |
| A.3.2 | Preferences |
| A.3.3 | Preferences and Profiles |
| A.3.4 | Display Filters |
| A.4 | Evaluation of RTP with Wireshark |
| A.4.1 | RTP Statistics |
| A.4.2 | RTP Stream Analysis |
| A.4.3 | Lab Exercise: RTP Basic Functions |
| A.4.4 | Lab Exercise: RTP Operation |
| A.5 | SIP Analysis with Wireshark |
| A.5.1 | VoIP Calls—Statistics |
| A.5.2 | SIP Statistics |
| A.5.3 | Lab Exercise: SIP Registration |
| A.5.4 | Lab Exercise: SIP—Basic Call with Wireshark |
| B | H.323—Terms and Procedures |
| B.1 | H.323 Architecture |
| B.1.1 | H.323 Procedures in the TCP/IP Protocol Stack |
| B.1.2 | H.323 Terminal—The Functions of the End Devices |
| B.1.3 | H.323 Gatekeeper—Address Translation and Management |
| B.1.4 | H.323 Gateway—The Translator |
| B.1.5 | The H.323 MCU—Conference Calls |
| B.2 | RAS—Services |
| B.2.1 | H.323 Identities and Addressing |
| B.2.2 | Registration with H.323 |
| B.2.3 | RAS Channel: Admission |
| B.3 | Call Setup and Termination |
| B.3.1 | H.225—Call Signaling Channel with Setup |
| B.3.2 | H.245 |
| B.3.3 | Call Termination |
| B.3.4 | A Complete Call |
| B.3.5 | H.323 Options: Fast Connect and Tunneling |
| B.4 | H.235—Encryption for H.323 |
| B.5 | H.323 and Firewalls |
| C | List of Abbreviations |
-
Classroom training
- Do you prefer the classic training method? A course in one of our Training Centers, with a competent trainer and the direct exchange between all course participants? Then you should book one of our classroom training dates!
-
Hybrid training
- Hybrid training means that online participants can additionally attend a classroom course. The dynamics of a real seminar are maintained, and the online participants are able to benefit from that. Online participants of a hybrid course use a collaboration platform, such as WebEx Training Center or Saba Meeting. To do this, a PC with browser and Internet access is required, as well as a headset and ideally a Web cam. In the seminar room, we use specially developed and customized audio- and video-technologies. This makes sure that the communication between all persons involved works in a convenient and fault-free way.
-
Online training
- You wish to attend a course in online mode? We offer you online course dates for this course topic. To attend these seminars, you need to have a PC with Internet access (minimum data rate 1Mbps), a headset when working via VoIP and optionally a camera. For further information and technical recommendations, please refer to.
-
Tailor-made courses
-
You need a special course for your team? In addition to our standard offer, we will also support you in creating your customized courses, which precisely meet your individual demands. We will be glad to consult you and create an individual offer for you.
Request for customized courses
You can find the complete description of this course with dates and prices ready for download at as PDF.
Guaranteed date
Classroom Training
Hybrid Training
Trainer language German