SIP protocol in VoIP Training in Yangon, Myanmar

This course provides an in-depth understanding of the Session Initiation Protocol (SIP), which is the backbone signaling protocol for managing voice, video, and messaging sessions in VoIP networks. Participants will explore the structure, operations, and functionalities of SIP in real-world VoIP applications. The course includes theoretical explanations, call flow scenarios, and hands-on labs using SIP tools and analyzers.

Course Agenda

Introduction to VoIP and SIP

• VoIP Overview and Architecture

• SIP's role in VoIP vs. H.323/MGCP

• Components: User Agents, Proxy, Registrar, Redirect Servers

SIP Protocol Essentials

• SIP Methods (INVITE, ACK, BYE, CANCEL, OPTIONS, REGISTER)

• SIP Message structure (Request/Response, Headers, Body)

• SIP URI and Addressing

SIP Call Flows

• Registration flow

• Basic call setup and teardown

• Call forwarding and transfer

• Forking and redirection

SDP and Media Negotiation

• Understanding SDP in SIP

• Codec negotiation

• Media stream handling with RTP

• SIP and QoS overview

SIP Tools and Lab Work

• Wireshark SIP filtering and packet analysis

• Using softphones (Zoiper, X-Lite)

• Setting up and testing calls with Asterisk or FreeSWITCH

• Scripting test calls with SIPp

SIP in NAT and Firewall Environments

• SIP NAT traversal problems

• STUN, TURN, ICE protocols

• SIP ALG issues

SIP Security

• Authentication and authorization

• TLS and SIP encryption

• SRTP and secure media

• Common vulnerabilities and countermeasures

Troubleshooting and Real-World Scenarios

• SIP call trace analysis

• Common errors (4xx, 5xx, 6xx responses)

• Debugging registration and media issues

• End-to-end scenario diagnosis

Hands-on Exercises