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Classic PBX access (BRI and PRI) for the connection of a company to the public voice network will be a thing of the past in a few years, In the course of the migration to Voice over IP, all PBX accesses therefore have to be replaced by an SIP trunk. In addition to the pure scaling of such an access, numerous technical details have to be clarified in the single case and challenges be mastered. The SIP Trunking course introduces the different connection scenarios with and without Enterprise Session Border Controller (E-SBC) and discusses the corresponding pros and cons. Redundancy concepts are just as important as ensuring voice quality and the transfer of fax messages. The frame stipulations of SIP Connect 1.1 are explained, which are meant to provide a homogenization. Hands-on exercises round off the course and illustrate the contents.
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Course Contents
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- Session Initiation Protocol
- SIPS (SIP over TLS)
- High Availability
- SIP Connect 1.1
- E-SBC
- Interoperability
- Quality of Service
- Fax over IP
- T.38
- (S)RTP
- Troubleshooting
- STUN, TURN, ICE
- Emergency Calls
- Features
- Call Number Administration
- Privacy Extensions
- IP Centrex
- VoIP Gateways
- Mobility
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.
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Target Group
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The course addresses persons already concerned with VoIP and SIP, who wish to acquire a better understanding of SIP trunks.
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Knowledge Prerequisites
-
Basic knowledge of VoIP and SIP will be helpful. This know-how can e.g. be acquired in the courses VoIP Fundamentals—SIP, H.323 & Co.in Application or SIP—The Universal Signaling Protocol.
| 1 State of affairs |
| 1.1 TC systems |
| 1.1.1 Control protocols |
| 1.2 System connection |
| 1.2.1 Features for DSS1 |
| 1.2.2 Performance features |
| 1.3 Voice as an application |
| 1.3.1 Voice coding and compression |
| 1.3.2 RTP transport |
| 1.3.3 RTCP - information about RTP connections |
| 1.4 IP Centrex |
| 1.5 The provider network |
| 1.5.1 User data |
| 1.5.2 Provider Access |
| 1.5.3 QoS in Provider Access |
| 1.5.4 Provider coupling |
| 2 SIP Trunking at a glance |
| 2.1 The basic principle of SIP trunking |
| 2.2 Connection variants |
| 2.2.1 Connection concepts without E-SBC |
| 2.2.2 Connection concepts with E-SBC |
| 2.2.3 Firewalls |
| 2.2.4 Encryption variants |
| 2.2.5 SIP trunking and NAT |
| 2.2.6 Redundancy concepts |
| 2.2.7 SIP trunk with classic PBX system |
| 2.2.8 Number blocks |
| 2.2.9 Distributed sites |
| 2.3 Interconnection |
| 2.3.1 Packet Switched to Packet Switched |
| 2.3.2 Packet Switched to Circuit Switched |
| 2.4 Provider without IMS |
| 3 Session Initiation Protocol |
| 3.1 The basic idea of SIP trunking |
| 3.1.1 The SIP message |
| 3.2 Registration and control |
| 3.2.1 Registration of a client |
| 3.2.2 Proxy Authentication |
| 3.3 A session in detail |
| 3.3.1 INVITE |
| 3.3.2 100 Trying |
| 3.3.3 180 Ringing and 183 Session Progress |
| 3.3.4 Disconnection and BYE - What to consider? |
| 3.4 SUBSCRIBE and NOTIFY |
| 3.5 OPTIONS |
| 3.6 PRACK - Reliable confirmation |
| 3.7 UPDATE - I still have one! |
| 3.8 REFER |
| 3.9 Session Description Protocol |
| 3.9.1 Structure of the Message Body with SDP |
| 3.9.2 SDP for advanced users |
| 3.9.3 RTP profiles |
| 3.10 Key tones |
| 4 SIP Trunking in detail |
| 4.1 Identity and Authentication |
| 4.1.1 Registration Mode and Static Mode |
| 4.1.2 Registration of groups |
| 4.1.3 P-Header Extensions on the SIP Trunk |
| 4.1.4 P-Preferred/ Asserted ID |
| 4.1.5 Authentication using P-Asserted-Identity |
| 4.1.6 P-Early-Media |
| 4.2 Session Border Controller |
| 4.2.1 SBC in detail |
| 4.2.2 Access control via session border controller |
| 4.3 Integration into the DMZ |
| 4.3.1 A network between networks |
| 4.3.2 Integration of the E-SBC into the DMZ |
| 4.4 NAT - Network Address Translation |
| 4.4.1 STUN tool for handling NAT |
| 4.4.2 Interactive Connectivity Establishment (ICE) |
| 4.4.3 SBC and NAT |
| 4.5 Securing signaling |
| 4.5.1 SIPS |
| 4.6 Securing the media stream |
| 4.6.1 SRTP and SRTCP packet formats |
| 4.6.2 Key management |
| 4.6.3 Key management for signaling |
| 4.6.4 Key management in Session Description Protocol |
| 4.6.5 DTLS-based key exchange |
| 4.7 Availability |
| 4.7.1 Load balancing methods (1) |
| 4.7.2 SIP in the domain environment |
| 4.8 Fax on the SIP trunk |
| 4.8.1 Special features of fax transmission |
| 4.8.2 Fax over IP - the possibilities |
| 4.8.3 The fax as a normal VoIP call |
| 4.8.4 T.37 - Fax as e-mail attachment |
| 4.8.5 T.38 - Fax in real time |
| 4.9 Features |
| 4.9.1 MMTel features |
| 4.9.2 Communication Diversion |
| 4.10 Emergency call |
| 5 SIPConnect |
| 5.1 SIP Trunk according to SIP Forum |
| 5.2 SIP Trunking Architecture |
| 5.3 Security |
| 5.4 Registration mode |
| 5.5 Static mode |
| 5.6 Enterprise public identities |
| 5.7 Handling of calls 1/2 |
| 5.7.1 P-Asserted-Identity |
| 5.7.2 Call transfer |
| 5.7.3 Media endpoints |
| 5.8 NAT & Firewalls |
| 5.9 Emergency call |
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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.-
Classic PBX access (BRI and PRI) for the connection of a company to the public voice network will be a thing of the past in a few years, In the course of the migration to Voice over IP, all PBX accesses therefore have to be replaced by an SIP trunk. In addition to the pure scaling of such an access, numerous technical details have to be clarified in the single case and challenges be mastered. The SIP Trunking course introduces the different connection scenarios with and without Enterprise Session Border Controller (E-SBC) and discusses the corresponding pros and cons. Redundancy concepts are just as important as ensuring voice quality and the transfer of fax messages. The frame stipulations of SIP Connect 1.1 are explained, which are meant to provide a homogenization. Hands-on exercises round off the course and illustrate the contents.
-
Course Contents
-
- Session Initiation Protocol
- SIPS (SIP over TLS)
- High Availability
- SIP Connect 1.1
- E-SBC
- Interoperability
- Quality of Service
- Fax over IP
- T.38
- (S)RTP
- Troubleshooting
- STUN, TURN, ICE
- Emergency Calls
- Features
- Call Number Administration
- Privacy Extensions
- IP Centrex
- VoIP Gateways
- Mobility
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 persons already concerned with VoIP and SIP, who wish to acquire a better understanding of SIP trunks.
-
Knowledge Prerequisites
-
Basic knowledge of VoIP and SIP will be helpful. This know-how can e.g. be acquired in the courses VoIP Fundamentals—SIP, H.323 & Co.in Application or SIP—The Universal Signaling Protocol.
| 1 State of affairs |
| 1.1 TC systems |
| 1.1.1 Control protocols |
| 1.2 System connection |
| 1.2.1 Features for DSS1 |
| 1.2.2 Performance features |
| 1.3 Voice as an application |
| 1.3.1 Voice coding and compression |
| 1.3.2 RTP transport |
| 1.3.3 RTCP - information about RTP connections |
| 1.4 IP Centrex |
| 1.5 The provider network |
| 1.5.1 User data |
| 1.5.2 Provider Access |
| 1.5.3 QoS in Provider Access |
| 1.5.4 Provider coupling |
| 2 SIP Trunking at a glance |
| 2.1 The basic principle of SIP trunking |
| 2.2 Connection variants |
| 2.2.1 Connection concepts without E-SBC |
| 2.2.2 Connection concepts with E-SBC |
| 2.2.3 Firewalls |
| 2.2.4 Encryption variants |
| 2.2.5 SIP trunking and NAT |
| 2.2.6 Redundancy concepts |
| 2.2.7 SIP trunk with classic PBX system |
| 2.2.8 Number blocks |
| 2.2.9 Distributed sites |
| 2.3 Interconnection |
| 2.3.1 Packet Switched to Packet Switched |
| 2.3.2 Packet Switched to Circuit Switched |
| 2.4 Provider without IMS |
| 3 Session Initiation Protocol |
| 3.1 The basic idea of SIP trunking |
| 3.1.1 The SIP message |
| 3.2 Registration and control |
| 3.2.1 Registration of a client |
| 3.2.2 Proxy Authentication |
| 3.3 A session in detail |
| 3.3.1 INVITE |
| 3.3.2 100 Trying |
| 3.3.3 180 Ringing and 183 Session Progress |
| 3.3.4 Disconnection and BYE - What to consider? |
| 3.4 SUBSCRIBE and NOTIFY |
| 3.5 OPTIONS |
| 3.6 PRACK - Reliable confirmation |
| 3.7 UPDATE - I still have one! |
| 3.8 REFER |
| 3.9 Session Description Protocol |
| 3.9.1 Structure of the Message Body with SDP |
| 3.9.2 SDP for advanced users |
| 3.9.3 RTP profiles |
| 3.10 Key tones |
| 4 SIP Trunking in detail |
| 4.1 Identity and Authentication |
| 4.1.1 Registration Mode and Static Mode |
| 4.1.2 Registration of groups |
| 4.1.3 P-Header Extensions on the SIP Trunk |
| 4.1.4 P-Preferred/ Asserted ID |
| 4.1.5 Authentication using P-Asserted-Identity |
| 4.1.6 P-Early-Media |
| 4.2 Session Border Controller |
| 4.2.1 SBC in detail |
| 4.2.2 Access control via session border controller |
| 4.3 Integration into the DMZ |
| 4.3.1 A network between networks |
| 4.3.2 Integration of the E-SBC into the DMZ |
| 4.4 NAT - Network Address Translation |
| 4.4.1 STUN tool for handling NAT |
| 4.4.2 Interactive Connectivity Establishment (ICE) |
| 4.4.3 SBC and NAT |
| 4.5 Securing signaling |
| 4.5.1 SIPS |
| 4.6 Securing the media stream |
| 4.6.1 SRTP and SRTCP packet formats |
| 4.6.2 Key management |
| 4.6.3 Key management for signaling |
| 4.6.4 Key management in Session Description Protocol |
| 4.6.5 DTLS-based key exchange |
| 4.7 Availability |
| 4.7.1 Load balancing methods (1) |
| 4.7.2 SIP in the domain environment |
| 4.8 Fax on the SIP trunk |
| 4.8.1 Special features of fax transmission |
| 4.8.2 Fax over IP - the possibilities |
| 4.8.3 The fax as a normal VoIP call |
| 4.8.4 T.37 - Fax as e-mail attachment |
| 4.8.5 T.38 - Fax in real time |
| 4.9 Features |
| 4.9.1 MMTel features |
| 4.9.2 Communication Diversion |
| 4.10 Emergency call |
| 5 SIPConnect |
| 5.1 SIP Trunk according to SIP Forum |
| 5.2 SIP Trunking Architecture |
| 5.3 Security |
| 5.4 Registration mode |
| 5.5 Static mode |
| 5.6 Enterprise public identities |
| 5.7 Handling of calls 1/2 |
| 5.7.1 P-Asserted-Identity |
| 5.7.2 Call transfer |
| 5.7.3 Media endpoints |
| 5.8 NAT & Firewalls |
| 5.9 Emergency call |
-
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
Hybrid Training
Trainer language German