WebRTC and SIP

Location: On-Site or Online
Pricing: $1,250 per seat (6-seat minimum)
Length: 4 Days

Course Summary

WebRTC and SIP is a practical, hands-on course designed for engineers who want to understand, deploy, and troubleshoot real-time communications systems using SIP signaling and WebRTC media technologies.

Students learn how voice and video calls actually work on the wire—from SIP registration and call setup to WebRTC media negotiation, NAT traversal, and secure transport. The course emphasizes protocol behavior, call flows, codecs, signaling vs media separation, and common failure modes encountered in production environments.

By the end of the course, students are comfortable reading call flows, debugging signaling and media issues, and understanding how modern browsers, softphones, SBCs, and VoIP systems interoperate.

Course Outline

Day 1 – Foundations of Real-Time Communications

• 💬 Lecture: Evolution of real-time communications (PSTN → VoIP → WebRTC)

• 💬 Lecture: High-level overview of SIP and WebRTC roles

• 💬 Lecture: Signaling vs media planes

• 💬 Lecture: Core terminology (UAC, UAS, proxy, registrar, peer connection)

• 💬 Lecture: Network fundamentals for real-time traffic

• ⚙️ Lab: Exploring SIP and WebRTC reference architectures

• ⚙️ Lab: Identifying signaling and media paths in a sample deployment

• ⚙️ Lab: Capturing VoIP traffic with packet analysis tools

• ⚙️ Lab: Differentiating signaling packets from media packets

Day 2 – SIP Protocol Deep Dive

• 💬 Lecture: SIP architecture and components

• 💬 Lecture: SIP messages, methods, and responses

• 💬 Lecture: SIP dialogs, transactions, and state

• 💬 Lecture: Registration, authentication, and location services

• 💬 Lecture: Session Description Protocol (SDP) fundamentals

• ⚙️ Lab: Registering a SIP endpoint

• ⚙️ Lab: Placing and receiving SIP calls

• ⚙️ Lab: Analyzing SIP INVITE, 100 Trying, 180 Ringing, 200 OK

• ⚙️ Lab: Inspecting SDP offers and answers

• ⚙️ Lab: Simulating common SIP failures (auth, routing, codec mismatch)

Day 3 – WebRTC Architecture and Media Flow

• 💬 Lecture: WebRTC architecture and browser security model

• 💬 Lecture: Peer connections, ICE, STUN, and TURN

• 💬 Lecture: Codec negotiation and media constraints

• 💬 Lecture: Secure media with DTLS and SRTP

• 💬 Lecture: WebRTC signaling (what WebRTC does not define)

• ⚙️ Lab: Establishing a basic WebRTC peer connection

• ⚙️ Lab: Inspecting ICE candidates and connection states

• ⚙️ Lab: Observing STUN and TURN behavior

• ⚙️ Lab: Analyzing RTP vs SRTP media streams

• ⚙️ Lab: Troubleshooting one-way audio and no-media scenarios

Day 4 – SIP + WebRTC Integration and Troubleshooting

• 💬 Lecture: SIP and WebRTC interworking models

• 💬 Lecture: WebRTC to SIP gateways and SBCs

• 💬 Lecture: NAT traversal challenges and solutions

• 💬 Lecture: Security considerations for signaling and media

• 💬 Lecture: Common real-world failure patterns

• ⚙️ Lab: Bridging a WebRTC client to a SIP endpoint

• ⚙️ Lab: Tracing a full call flow (browser → SIP → media)

• ⚙️ Lab: Debugging failed calls using logs and packet captures

• ⚙️ Lab: Identifying signaling success with media failure

• ⚙️ Lab: End-to-end troubleshooting workflow exercise

Outcomes

Students who complete WebRTC and SIP will be able to:

• Explain how SIP and WebRTC calls work end to end

• Read and interpret SIP call flows and SDP exchanges

• Understand WebRTC media negotiation and NAT traversal

• Distinguish signaling issues from media path failures

• Troubleshoot real-world VoIP and WebRTC problems with confidence