The Session Initiation Protocol (SIP) has become the preferred signaling protocol both for Voice over IP (VoIP) and Next Generation Networks. The advantages of SIP are simple processes and interesting applications. One major advantage of SIP is that it can be easily extended: New formats are no problem, both synchronous and asynchronous data streams can be initiated, and the communications partners can have a peer-to-peer or client-server relationship. Due to the introduction of VoIP in public voice networks, SIP has improved significantly.
Course Contents
• VoIP Data Streams (RTP, RTCP, and Signaling)
• User Agent (UA), SIP Proxy Server, Registrar, SIP Redirect Server, and Location Service
• SIP Protocol: Message Types and Their Setup
• SIP URIs and TEL URIs Address Formats and Their Application
• SDP Protocol: Contents and Tasks
• RTP/AVT Profiles
• Typical SIP Processes during Connection Setup and a SIP Call
• Features
• Interaction of SIP with NAT and Firewalls
• Fax with T.38
• SIP and UMTS-the IP Multimedia Subsystem (IMS)
• Application of SIP in Provider Networks
• Back-to-Back User Agent (B2BUA) and Session Border Controller (SBC)
Detailed Table of Contents
Practice-related presentations and the analysis of traces will make the course contents easier to understand.
In this course from the ExperTeach Networking series, each participant will receive the comprehensive ExperTeach course documentation.
Target Group
Network designers belong to the target group of the course as much as employees who have to be familiar with SIP on the protocol layer.
Knowledge Prerequisites
A well-based know-how of the voice and IP sectors is the prerequisite for this course. Basic VoIP know-how will be very helpful for course participants.
Course Objective
After the course, the students will know the advantages, special features, and application options of the SIP architecture. The knowledge of the corresponding protocol will help to enhance the overall understanding.
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