-
Voice over IP is an application that is particularly prone to errors due to its special requirements. Special skills and tools are required to analyze problems with connection setup, connection stability or voice quality. A VoIP environment also uses a variety of standardized and manufacturer-specific signaling protocols for voice, video and instant messaging. In this course, participants will learn about the functions and analysis of the most important VoIP processes and protocols with the help of Wireshark. One focus is on the practical detection of typical problems.
-
Course Contents
-
- Overview of VoIP - motivation and basics
- VoIP - standards, components and protocols
- Media streams - functions, codecs, RTP and RTCP
- Call signaling with Wireshark - SIP, Skinny and H.323
- Wireshark evaluations for VoIP
- Practical analysis of SIP - registration, authentication, connection setup
- Performance features with SIP - forwarding, conference, instant messaging, etc.
- Analysis of dropped calls and voice quality with Wireshark
- Fax over IP - How it works!
- Quality of service for VoIP applications
- Analysis of problems with NAT and firewalls
The detailed digital documentation package, consisting of an e-book and PDF, is included in the price of the course.
Premium Course Documents
In addition to the digital documentation package, the exclusive Premium Print Package is also available to you.
- High-quality color prints of the ExperTeach documentation
- Exclusive folder in an elegant design
- Document pouch in backpack shape
- Elegant LAMY ballpoint pen
- Practical notepad
The Premium Print Package can be added during the ordering process for € 150,- plus VAT (only for classroom participation).
-
Target Group
-
The course is designed for networkers who are responsible for the operation and troubleshooting of networks for VoIP and want to analyze VoIP applications with the help of Wireshark.
-
Knowledge Prerequisites
-
Participants should have sound practical experience in using Wireshark as well as knowledge of TCP/IP and IP addressing. Prior attendance of the basic course Wireshark Protocol Analysis – Practical Use in the Network is highly recommended.
| 1 | Motivation and Basics |
| 1.1 | Voice Networks of Today and Tomorrow |
| 1.2 | VoIP Architecture |
| 1.2.1 | VoIP—Requirements |
| 1.2.2 | Data Streams |
| 1.2.3 | VoIP Protocols |
| 1.2.4 | Signaling |
| 1.2.5 | Media Streams |
| 1.3 | Conferencing and WebRTC |
| 1.3.1 | WebRTC—The Open Conferencing Solution |
| 1.3.2 | Browser or Apps |
| 1.4 | Wireshark in a Brief Overview |
| 1.4.1 | Capture Options |
| 1.4.2 | Capture Filter |
| 1.4.3 | Preferences |
| 1.4.4 | Preferences and Profiles |
| 1.4.5 | Display Filters |
| 2 | VoIP Basics—RTP and SIP |
| 2.1 | The Real-time Transport Protocol |
| 2.1.1 | RTP Packets and Protocol Headers |
| 2.1.2 | RTP Profiles |
| 2.2 | Real-time Transport Control Protocol (RTCP) |
| 2.2.1 | RTCP Sender and Receiver Report |
| 2.2.2 | RTCP Extended Reports (RTCP XR) |
| 2.3 | Measuring Voice Quality |
| 2.3.1 | Mean Opinion Score (MOS) |
| 2.3.2 | Delays—End-to-End |
| 2.3.3 | Jitter and Jitter Buffer |
| 2.3.4 | Packet Loss and Concealment |
| 2.4 | SIP—An Overview |
| 2.4.1 | Standardization |
| 2.4.2 | Addressing: SIP URI and TEL URI |
| 2.4.3 | The Components of the SIP Architecture |
| 2.5 | SIP Protocol Setup and Processes |
| 2.5.1 | Setup of SIP Messages |
| 2.5.2 | SIP Requests—SIP Methods |
| 2.5.3 | SIP Responses |
| 2.6 | SDP—The Session Description Protocol |
| 2.6.1 | SDP in Call Setup |
| 2.6.2 | Modifying an Existing Connection |
| 2.7 | Registration and Authentication |
| 2.7.1 | SIP Registration—Processes |
| 2.7.2 | Registration without Authentication |
| 2.7.3 | Register with Authentication |
| 2.8 | Basic Function—Basic Call |
| 2.8.1 | SIP Invite over the Classic Proxy |
| 2.8.2 | SIP Server Terminates the Dialog |
| 2.8.3 | Domain Environments and DNS |
| 3 | VoIP Analysis with Wireshark |
| 3.1 | Measurement Technology for VoIP |
| 3.1.1 | Wireshark Measurement on End Devices |
| 3.1.2 | External Measuring Tools |
| 3.1.3 | VoIP Simulation and VoIP Tests |
| 3.1.4 | Measurement on the PBX or SBC |
| 3.2 | Evaluation of RTP with Wireshark |
| 3.2.1 | Capturing RTP with Wireshark |
| 3.2.2 | Decode RTP |
| 3.2.3 | RTP Statistics |
| 3.2.4 | RTP Stream Analysis |
| 3.3 | SIP Analysis with Wireshark |
| 3.3.1 | SIP—Useful Filters |
| 3.3.2 | VoIP Calls—Statistics |
| 3.3.3 | SIP Statistics |
| 3.4 | Encryption of SIP and RTP |
| 3.4.1 | SRTP |
| 3.4.2 | SRTP and Wireshark |
| 3.4.3 | SIPS—SIP over TLS |
| 3.4.4 | SIPS Decoding in Wireshark |
| 4 | VoIP Analysis and Troubleshooting |
| 4.1 | Limitation of the Problem |
| 4.2 | VoIP Analysis—Voice Problems |
| 4.2.1 | Dead Air—No Voice |
| 4.2.2 | RTP—One-sided Communication |
| 4.2.3 | Problems with Codecs |
| 4.2.4 | Problems with Encryption |
| 4.3 | VoIP Analysis—Registration |
| 4.3.1 | Find Registration Problems with Wireshark |
| 4.3.2 | Multiple Registration |
| 4.3.3 | SIPS and Certificate Problems |
| 4.4 | Problems during Connection Setup |
| 4.4.1 | No Connection Setup |
| 4.4.2 | Slow Connection Setup |
| 4.4.3 | No Ringing Tone during Call Setup |
| 4.5 | Problems During the Connection |
| 4.5.1 | Connection Breakdown |
| 5 | VoIP—Practical Topics |
| 5.1 | VoIP and Stateful Firewalls |
| 5.1.1 | Application Layer Gateway |
| 5.1.2 | NAT—Network Address Translation |
| 5.2 | Fax over IP—That’s how it works! |
| 5.2.1 | Special Features during Fax Transmission |
| 5.2.2 | Typical Procedures |
| 5.2.3 | Fax as a Default VoIP Call |
| 5.2.4 | T.37—Fax as e-Mail Attachment |
| 5.2.5 | T.38—Fax in Real-Time |
| 5.2.6 | Error Patterns with Fax over IP |
| 5.3 | Quality of Service in View |
| 5.3.1 | QoS Concepts |
| 5.3.2 | QoS in the LAN |
| 5.3.3 | DiffServ |
| 5.3.4 | Check QoS with Wireshark |
| 5.4 | WebRTC—The Open Conferencing Solution |
| 5.4.1 | WebRTC Standardization |
| 5.4.2 | WebRTC Architecture |
| 5.4.3 | WebRTC Protocol Stacks |
| 5.4.4 | Control Tasks during the Setup of WebRTC |
| 5.4.5 | Media Streams |
| 5.4.6 | WebRTC Example in Wireshark |
| A | Lab Exercises and Solutions |
| A.1 | Lab Exercises in the Course |
| A.1.1 | Virtual Lab |
| A.1.2 | Preparations on the Client PC |
| A.1.3 | Configuration of VoIP Clients |
| A.1.4 | VoIP PBX |
| A.2 | Lab Exercises—RTP and SIP Basics |
| A.2.1 | RTP Basic Functions |
| A.2.2 | RTP Operation |
| A.2.3 | SIP Registration |
| A.2.4 | SIP Registration at the SIP Trunk |
| A.2.5 | SIP—Basic Call |
| A.2.6 | Secure RTP |
| A.2.7 | SIP over TLS |
| A.3 | Lab Exercise—SIP Lab |
| A.3.1 | Registration and Basic Call |
| A.3.2 | Selection of the Codec |
| A.3.3 | RTP Proxy |
| A.3.4 | Call Hold |
| A.3.5 | Simple Three-Party Conference |
| A.3.6 | Call Transfer |
| A.4 | Lab Exercises—Problems with Voice Quality |
| A.4.1 | Interference with Voice Quality |
| A.4.2 | Poor Voice Quality over the WAN Link |
| A.4.3 | Poor Voice Quality at One End |
| A.4.4 | Even More Voice Problems |
| A.4.5 | No Voice over Network–1 |
| A.4.6 | No Voice over Network–2 |
| A.5 | Lab Exercises—Connection Problems with SIP |
| A.5.1 | Connection Problems: Case 1 |
| A.5.2 | Faulty Trace |
| A.5.3 | SIP Multiple Registration |
| A.5.4 | SIP Timer—1 |
| A.5.5 | SIP Timer—2 |
| A.5.6 | SIP Problems on the Trunk |
| A.6 | Solutions to the Lab Exercises |
| A.6.1 | Solutions to the Lab Exercises—Basics |
| A.6.2 | Solutions to the Lab Exercises—Problems with Voice Quality |
| A.6.3 | Solutions to the Lab Exercises—Connection Problems |
| B | RTP Functions |
| B.1 | DTMF—Dial Tones via VoIP |
| B.1.1 | DTMF Inband |
| B.1.2 | RTP Event According to RFC 4733 (RFC 2833) |
| B.1.3 | DTMF via SIP Info |
| B.2 | Speech Pauses and VAD |
| B.2.1 | Speech Pauses and RTP |
| B.2.2 | VAD Indicators in Wireshark |
| B.2.3 | Comfort Noise |
| B.3 | SIP Response Codes |
| 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!
-
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.
You can find the complete description of this course with dates and prices ready for download at as PDF.-
Voice over IP is an application that is particularly prone to errors due to its special requirements. Special skills and tools are required to analyze problems with connection setup, connection stability or voice quality. A VoIP environment also uses a variety of standardized and manufacturer-specific signaling protocols for voice, video and instant messaging. In this course, participants will learn about the functions and analysis of the most important VoIP processes and protocols with the help of Wireshark. One focus is on the practical detection of typical problems.
-
Course Contents
-
- Overview of VoIP - motivation and basics
- VoIP - standards, components and protocols
- Media streams - functions, codecs, RTP and RTCP
- Call signaling with Wireshark - SIP, Skinny and H.323
- Wireshark evaluations for VoIP
- Practical analysis of SIP - registration, authentication, connection setup
- Performance features with SIP - forwarding, conference, instant messaging, etc.
- Analysis of dropped calls and voice quality with Wireshark
- Fax over IP - How it works!
- Quality of service for VoIP applications
- Analysis of problems with NAT and firewalls
The detailed digital documentation package, consisting of an e-book and PDF, is included in the price of the course.
Premium Course Documents
In addition to the digital documentation package, the exclusive Premium Print Package is also available to you.
- High-quality color prints of the ExperTeach documentation
- Exclusive folder in an elegant design
- Document pouch in backpack shape
- Elegant LAMY ballpoint pen
- Practical notepad
The Premium Print Package can be added during the ordering process for € 150,- plus VAT (only for classroom participation).
-
Target Group
-
The course is designed for networkers who are responsible for the operation and troubleshooting of networks for VoIP and want to analyze VoIP applications with the help of Wireshark.
-
Knowledge Prerequisites
-
Participants should have sound practical experience in using Wireshark as well as knowledge of TCP/IP and IP addressing. Prior attendance of the basic course Wireshark Protocol Analysis – Practical Use in the Network is highly recommended.
| 1 | Motivation and Basics |
| 1.1 | Voice Networks of Today and Tomorrow |
| 1.2 | VoIP Architecture |
| 1.2.1 | VoIP—Requirements |
| 1.2.2 | Data Streams |
| 1.2.3 | VoIP Protocols |
| 1.2.4 | Signaling |
| 1.2.5 | Media Streams |
| 1.3 | Conferencing and WebRTC |
| 1.3.1 | WebRTC—The Open Conferencing Solution |
| 1.3.2 | Browser or Apps |
| 1.4 | Wireshark in a Brief Overview |
| 1.4.1 | Capture Options |
| 1.4.2 | Capture Filter |
| 1.4.3 | Preferences |
| 1.4.4 | Preferences and Profiles |
| 1.4.5 | Display Filters |
| 2 | VoIP Basics—RTP and SIP |
| 2.1 | The Real-time Transport Protocol |
| 2.1.1 | RTP Packets and Protocol Headers |
| 2.1.2 | RTP Profiles |
| 2.2 | Real-time Transport Control Protocol (RTCP) |
| 2.2.1 | RTCP Sender and Receiver Report |
| 2.2.2 | RTCP Extended Reports (RTCP XR) |
| 2.3 | Measuring Voice Quality |
| 2.3.1 | Mean Opinion Score (MOS) |
| 2.3.2 | Delays—End-to-End |
| 2.3.3 | Jitter and Jitter Buffer |
| 2.3.4 | Packet Loss and Concealment |
| 2.4 | SIP—An Overview |
| 2.4.1 | Standardization |
| 2.4.2 | Addressing: SIP URI and TEL URI |
| 2.4.3 | The Components of the SIP Architecture |
| 2.5 | SIP Protocol Setup and Processes |
| 2.5.1 | Setup of SIP Messages |
| 2.5.2 | SIP Requests—SIP Methods |
| 2.5.3 | SIP Responses |
| 2.6 | SDP—The Session Description Protocol |
| 2.6.1 | SDP in Call Setup |
| 2.6.2 | Modifying an Existing Connection |
| 2.7 | Registration and Authentication |
| 2.7.1 | SIP Registration—Processes |
| 2.7.2 | Registration without Authentication |
| 2.7.3 | Register with Authentication |
| 2.8 | Basic Function—Basic Call |
| 2.8.1 | SIP Invite over the Classic Proxy |
| 2.8.2 | SIP Server Terminates the Dialog |
| 2.8.3 | Domain Environments and DNS |
| 3 | VoIP Analysis with Wireshark |
| 3.1 | Measurement Technology for VoIP |
| 3.1.1 | Wireshark Measurement on End Devices |
| 3.1.2 | External Measuring Tools |
| 3.1.3 | VoIP Simulation and VoIP Tests |
| 3.1.4 | Measurement on the PBX or SBC |
| 3.2 | Evaluation of RTP with Wireshark |
| 3.2.1 | Capturing RTP with Wireshark |
| 3.2.2 | Decode RTP |
| 3.2.3 | RTP Statistics |
| 3.2.4 | RTP Stream Analysis |
| 3.3 | SIP Analysis with Wireshark |
| 3.3.1 | SIP—Useful Filters |
| 3.3.2 | VoIP Calls—Statistics |
| 3.3.3 | SIP Statistics |
| 3.4 | Encryption of SIP and RTP |
| 3.4.1 | SRTP |
| 3.4.2 | SRTP and Wireshark |
| 3.4.3 | SIPS—SIP over TLS |
| 3.4.4 | SIPS Decoding in Wireshark |
| 4 | VoIP Analysis and Troubleshooting |
| 4.1 | Limitation of the Problem |
| 4.2 | VoIP Analysis—Voice Problems |
| 4.2.1 | Dead Air—No Voice |
| 4.2.2 | RTP—One-sided Communication |
| 4.2.3 | Problems with Codecs |
| 4.2.4 | Problems with Encryption |
| 4.3 | VoIP Analysis—Registration |
| 4.3.1 | Find Registration Problems with Wireshark |
| 4.3.2 | Multiple Registration |
| 4.3.3 | SIPS and Certificate Problems |
| 4.4 | Problems during Connection Setup |
| 4.4.1 | No Connection Setup |
| 4.4.2 | Slow Connection Setup |
| 4.4.3 | No Ringing Tone during Call Setup |
| 4.5 | Problems During the Connection |
| 4.5.1 | Connection Breakdown |
| 5 | VoIP—Practical Topics |
| 5.1 | VoIP and Stateful Firewalls |
| 5.1.1 | Application Layer Gateway |
| 5.1.2 | NAT—Network Address Translation |
| 5.2 | Fax over IP—That’s how it works! |
| 5.2.1 | Special Features during Fax Transmission |
| 5.2.2 | Typical Procedures |
| 5.2.3 | Fax as a Default VoIP Call |
| 5.2.4 | T.37—Fax as e-Mail Attachment |
| 5.2.5 | T.38—Fax in Real-Time |
| 5.2.6 | Error Patterns with Fax over IP |
| 5.3 | Quality of Service in View |
| 5.3.1 | QoS Concepts |
| 5.3.2 | QoS in the LAN |
| 5.3.3 | DiffServ |
| 5.3.4 | Check QoS with Wireshark |
| 5.4 | WebRTC—The Open Conferencing Solution |
| 5.4.1 | WebRTC Standardization |
| 5.4.2 | WebRTC Architecture |
| 5.4.3 | WebRTC Protocol Stacks |
| 5.4.4 | Control Tasks during the Setup of WebRTC |
| 5.4.5 | Media Streams |
| 5.4.6 | WebRTC Example in Wireshark |
| A | Lab Exercises and Solutions |
| A.1 | Lab Exercises in the Course |
| A.1.1 | Virtual Lab |
| A.1.2 | Preparations on the Client PC |
| A.1.3 | Configuration of VoIP Clients |
| A.1.4 | VoIP PBX |
| A.2 | Lab Exercises—RTP and SIP Basics |
| A.2.1 | RTP Basic Functions |
| A.2.2 | RTP Operation |
| A.2.3 | SIP Registration |
| A.2.4 | SIP Registration at the SIP Trunk |
| A.2.5 | SIP—Basic Call |
| A.2.6 | Secure RTP |
| A.2.7 | SIP over TLS |
| A.3 | Lab Exercise—SIP Lab |
| A.3.1 | Registration and Basic Call |
| A.3.2 | Selection of the Codec |
| A.3.3 | RTP Proxy |
| A.3.4 | Call Hold |
| A.3.5 | Simple Three-Party Conference |
| A.3.6 | Call Transfer |
| A.4 | Lab Exercises—Problems with Voice Quality |
| A.4.1 | Interference with Voice Quality |
| A.4.2 | Poor Voice Quality over the WAN Link |
| A.4.3 | Poor Voice Quality at One End |
| A.4.4 | Even More Voice Problems |
| A.4.5 | No Voice over Network–1 |
| A.4.6 | No Voice over Network–2 |
| A.5 | Lab Exercises—Connection Problems with SIP |
| A.5.1 | Connection Problems: Case 1 |
| A.5.2 | Faulty Trace |
| A.5.3 | SIP Multiple Registration |
| A.5.4 | SIP Timer—1 |
| A.5.5 | SIP Timer—2 |
| A.5.6 | SIP Problems on the Trunk |
| A.6 | Solutions to the Lab Exercises |
| A.6.1 | Solutions to the Lab Exercises—Basics |
| A.6.2 | Solutions to the Lab Exercises—Problems with Voice Quality |
| A.6.3 | Solutions to the Lab Exercises—Connection Problems |
| B | RTP Functions |
| B.1 | DTMF—Dial Tones via VoIP |
| B.1.1 | DTMF Inband |
| B.1.2 | RTP Event According to RFC 4733 (RFC 2833) |
| B.1.3 | DTMF via SIP Info |
| B.2 | Speech Pauses and VAD |
| B.2.1 | Speech Pauses and RTP |
| B.2.2 | VAD Indicators in Wireshark |
| B.2.3 | Comfort Noise |
| B.3 | SIP Response Codes |
| 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!
-
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.
You can find the complete description of this course with dates and prices ready for download at as PDF.