44  IoT Real-Time Protocols

In 60 Seconds

Real-time IoT protocols (RTP, SIP, WebRTC) sacrifice reliability for timing, using UDP to avoid retransmission delays that would cause stuttering video or garbled audio. For IoT devices like smart doorbells and intercoms, the 150ms latency threshold determines whether to use real-time streaming protocols or standard messaging protocols like MQTT and CoAP.

44.1 Learning Objectives

By the end of this chapter, you will be able to:

  • Explain VoIP Architecture: Describe the complete protocol stack (SIP, RTP, SRTP, RTCP) for real-time audio/video in IoT devices
  • Compare RTP vs Messaging Protocols: Distinguish when to use RTP instead of MQTT/CoAP based on latency, packet-loss tolerance, and data type
  • Design Secure Streaming: Configure encryption (SRTP, TLS) and authentication for real-time IoT streams accessible from the internet
  • Select Appropriate Protocols: Justify the choice between SIP, RTSP, and WebRTC for specific IoT device types and deployment scenarios
  • Calculate Latency Budgets: Apply the 150ms end-to-end threshold to evaluate whether a proposed protocol stack meets real-time conversation requirements
  • Analyze Hybrid Architectures: Assess when to combine messaging protocols (MQTT) with real-time protocols (RTP/SIP) in a single IoT system

Some IoT applications cannot tolerate any delay – a self-driving car needs to react in milliseconds, not seconds. Real-time protocols are designed to guarantee that data arrives on time, every time. Think of them as an express lane on a highway where critical traffic always gets through without waiting.

Most Valuable Understanding (MVU)

Real-time IoT protocols (RTP/SIP/WebRTC) prioritize latency over reliability: they run over UDP, skip lost packets rather than retransmit, and require <150ms end-to-end delay for acceptable human perception. Use them for continuous audio/video streams (doorbells, intercoms), but stick with MQTT/CoAP for discrete sensor data.

This is the fundamental trade-off: messaging protocols (MQTT, CoAP) guarantee delivery but accept latency, while real-time protocols (RTP) guarantee timing but accept packet loss. A video doorbell that retransmits lost packets would have unwatchable, stuttering video. Instead, it skips the lost frame and continues - the human eye interpolates the gap.

The 150ms Rule: Human conversation becomes awkward when one-way end-to-end delay exceeds 150ms (ITU-T G.114). This is why real-time IoT devices use UDP (no TCP handshake overhead) and SIP/RTP (designed for media timing) instead of HTTP/MQTT.

44.2 Prerequisites

Before diving into this chapter, you should be familiar with:


Hey there, future IoT engineer! Let’s learn about real-time protocols with the Sensor Squad!

Sammy the Sensor just got a new job: being the brain of a smart video doorbell! But there’s a problem - when someone rings the doorbell, Sammy needs to send live video to your phone SUPER fast!

The Big Difference:

Lila the Light sends temperature readings using MQTT. If a message gets lost, no big deal - another one comes in a minute! It’s like sending postcards: “It’s 72 degrees!” … (lost in mail) … “It’s 73 degrees!” You didn’t miss much!

But Sammy’s video doorbell is different. Imagine if your video call worked like postcards: - “Hi, I’m at your door–” (lost) - “–nd I have a package–” (lost) - “–ease open up!” (arrived)

You’d only hear “ease open up!” That makes no sense!

The Magic of RTP (Real-time Transport Protocol):

Instead of waiting for lost messages like MQTT does, RTP uses a different magic trick:

Protocol Lost Message Strategy Why
MQTT Wait and resend “I MUST deliver this temperature!”
RTP Skip it, keep going “Show the next video frame instead!”

Max the Motor explains: “When you’re watching a video, your eyes are SO fast that if we skip one tiny frame (1/30th of a second), you don’t even notice! But if we WAITED to resend it, you’d see the video freeze - yuck!”

The 150 Millisecond Rule:

Bella the Button discovered something cool: humans get confused if a phone call has more than 150 milliseconds (0.15 seconds) of delay!

  • Under 150ms: “Hello!” … “Hi there!” (normal conversation)
  • Over 300ms: “Hello!” … … … “Hi there!” (awkward, talking over each other)

That’s why video doorbells use UDP (super fast, no waiting) instead of TCP (reliable, but slow)!

Real-World Doorbell Flow:

Kid-friendly sequence diagram showing how a video doorbell works. A visitor presses the button, MQTT sends a notification to the phone app, SIP starts a video call, RTP streams live video at 30 frames per second, the user talks back through the speaker, and when the call ends, MQTT sends a 'video saved' notification.

Fun Fact: Your favorite video doorbell uses BOTH types of protocols! MQTT for notifications (needs to arrive), RTP for live video (needs to be fast). It’s like having a postcard for important news AND a phone call for live conversations!

Try This: Next time you use a video call, notice how sometimes a face looks “blocky” for a second, then gets clear again. That’s the protocol skipping some lost data and catching up - just like Sammy’s doorbell!

44.3 Real-time Protocols for IoT

While MQTT and CoAP handle most IoT data exchange scenarios, some applications require real-time audio and video streaming with strict latency requirements. Video doorbells, baby monitors, voice assistants, and intercom systems all need protocols designed specifically for continuous media streams.

44.3.1 IoT Protocol Ecosystem: Where Real-Time Fits

Comprehensive IoT protocol ecosystem diagram showing three main categories: Messaging Protocols (MQTT, CoAP, HTTP, AMQP) for sensor data and commands, Real-Time Protocols (SIP, RTP, SRTP, WebRTC, RTSP) for audio and video streaming, and Hybrid Approaches where devices use both types. The diagram highlights that video doorbells and smart displays typically use both protocol families for different purposes.

This ecosystem view shows that modern IoT devices often use both messaging and real-time protocols - each optimized for its specific purpose.

44.3.2 Real-Time Protocol Decision Flow

Decision flowchart for selecting IoT protocols. Starting with 'What type of data?', if sensor/telemetry, use MQTT or CoAP. If audio/video streaming, check if two-way communication is needed. Two-way leads to SIP+RTP for doorbells and intercoms. One-way leads to RTSP+RTP for cameras and monitors. If browser compatibility is needed, use WebRTC. All real-time paths require less than 150ms latency.

44.4 VoIP and SIP Architecture

Voice over IP (VoIP) enables real-time voice and video communication over IP networks. The protocol stack for VoIP consists of several layers working together:

44.4.1 SIP Call Flow for Video Doorbell

Sequence diagram showing SIP call flow between a video doorbell, SIP server, and mobile phone app. The doorbell sends an INVITE to the SIP server, which forwards it to the phone. After 200 OK acknowledgments, RTP/SRTP media streams flow directly between the doorbell and phone. When the call ends, a BYE message terminates the session.

VoIP protocol stack diagram showing layered architecture with SIP for session control at the top, RTP/RTCP for media transport in the middle running over UDP, and security layers including SRTP for encrypted media and TLS for encrypted signaling. The stack shows how these protocols work together to enable secure real-time audio and video communication in IoT devices like video doorbells and intercoms.

VoIP Protocol Stack: SIP Signaling with RTP Media Transport
Figure 44.1: VoIP Protocol Stack: SIP handles session control (call setup/teardown), while RTP carries the actual audio/video data over UDP. Security layers (SRTP, TLS, zRTP) protect both signaling and media streams.

44.5 Key Protocol Components

Protocol RFC Purpose IoT Relevance
SIP RFC 3261 Session Initiation Protocol - multimedia session control Call setup for video doorbells, intercoms
RTP RFC 3550 Real-time Transport Protocol - media stream delivery Audio/video streaming from cameras
RTCP RFC 3550 RTP Control Protocol - quality monitoring Adaptive bitrate for constrained networks
UDP RFC 768 User Datagram Protocol - connectionless transport Low-latency delivery (no TCP handshake)
SRTP RFC 3711 Secure RTP - encrypted media Privacy for baby monitors, doorbells
zRTP RFC 6189 Key exchange for SRTP End-to-end encryption setup
TLS RFC 8446 (TLS 1.3) Transport Layer Security - encrypted signaling Secure SIP (SIPS) on port 5061

44.6 SIP Ports and Security

SIP Port Assignments
Port Protocol Security Use Case
5060 SIP over UDP/TCP Unencrypted Internal/trusted networks
5061 SIPS over TLS Encrypted Internet-facing devices

For IoT devices accessible from the internet (video doorbells, remote intercoms), always use port 5061 with TLS encryption to prevent eavesdropping and session hijacking.

44.7 Real-time IoT Applications

Diagram showing real-time protocol selection for different IoT device categories. Two-way communication devices like video doorbells and smart intercoms use SIP plus RTP or SRTP. One-way streaming devices like IP cameras and baby monitors use RTSP plus RTP. Browser-based devices like voice assistants and smart displays use WebRTC. All paths converge on the requirement for sub-150ms latency and UDP-based transport.

Real-Time IoT Device Protocol Selection
Figure 44.2: Real-time Protocol Selection for IoT: Two-way communication devices (doorbells, intercoms) use SIP+RTP, while one-way streaming devices (cameras, monitors) often use RTSP+RTP. Voice assistants increasingly use WebRTC for browser compatibility.

Common IoT Use Cases:

Device Protocol Why
Video Doorbell SIP + SRTP Two-way audio/video with visitor, needs secure encrypted stream
Baby Monitor RTP (or RTSP) One-way video stream, low latency critical for responsiveness
Smart Intercom SIP + RTP Full duplex audio between rooms, session-based communication
Voice Assistant WebRTC or proprietary Browser/app integration, cloud speech processing
Security Camera RTSP + RTP Continuous streaming, ONVIF standard interoperability

44.8 WebRTC for Browser-Based IoT

WebRTC (Web Real-Time Communication) has become increasingly important for IoT devices that need browser-based access. It provides peer-to-peer audio/video streaming with built-in NAT traversal and encryption.

44.8.1 WebRTC Connection Architecture

WebRTC connection architecture showing an IoT device (smart display) connecting to a web browser through STUN and TURN servers for NAT traversal. The signaling server exchanges SDP offers and answers. Once connected, peer-to-peer SRTP media flows directly between device and browser, with TURN relay as fallback when direct connection fails.

WebRTC Components:

Component Purpose IoT Use Case
getUserMedia Access camera/microphone Smart display video call
RTCPeerConnection Establish P2P connection Direct device-to-browser stream
STUN Discover public IP/port NAT traversal for home networks
TURN Relay when direct fails Symmetric NAT, firewall scenarios
DTLS-SRTP Encrypted media Secure by default
WebRTC vs SIP for IoT

Choose WebRTC when:

  • Browser access required (no app installation)
  • NAT traversal is complex (STUN/TURN built-in)
  • You want encryption by default

Choose SIP + RTP when:

  • Integrating with existing VoIP infrastructure
  • Need PSTN gateway (phone network)
  • SIP-based intercom systems

44.9 Comparison: Real-time vs Messaging Protocols

When should you use VoIP/RTP instead of MQTT/CoAP?

44.9.1 Protocol Architecture Comparison

Side-by-side comparison of messaging protocols (MQTT/CoAP) versus real-time protocols (RTP/SIP). Left side shows MQTT/CoAP characteristics: discrete messages, TCP/UDP transport, retransmission on loss, low bandwidth, and typical payloads of JSON/CBOR sensor data. Right side shows RTP/SIP characteristics: continuous streams, UDP transport, skip lost packets, high bandwidth, and typical payloads of H.264 video and PCM audio. Center shows the key trade-off: reliability vs latency.

Requirement MQTT/CoAP VoIP/RTP
Data Type Sensor readings, commands, telemetry Continuous audio/video streams
Latency Tolerance 100ms - seconds acceptable <150ms required (human perception)
Packet Loss Retransmit (reliability critical) Skip/interpolate (continuity critical)
Bandwidth Low (bytes to KB per message) High (64kbps - 2Mbps continuous)
Connection Model Message-based (discrete) Session-based (continuous stream)
Typical Payload JSON, CBOR, binary sensor data PCM audio, H.264/H.265 video

Use RTP/SRTP when:

  • Streaming continuous audio/video (doorbell live view)
  • Two-way voice communication (intercom, doorbell talk)
  • Latency under 150ms is critical
  • You need synchronized audio/video playback

Use MQTT when:

  • Sending audio clips/recordings (doorbell motion event)
  • Push notifications with audio alert
  • Speech-to-text results from cloud processing
  • Audio metadata (noise level, voice detection events)

Hybrid approach (common in smart doorbells):

Motion detected → MQTT notification to phone app
User opens app → SIP session established
Live video/audio → RTP/SRTP stream
User speaks → RTP audio to doorbell speaker
Call ends → SIP session terminated
Event recorded → Video clip stored, MQTT notification

44.10 Security for Real-time IoT

Real-time audio and video streams require strong security to prevent eavesdropping:

44.10.1 Security Layer Architecture

Layered security architecture for real-time IoT showing four security layers: Authentication layer with digest auth and certificates at the top, Key Exchange layer with zRTP and DTLS-SRTP in the middle, Media Encryption layer with SRTP below that, and Signaling Encryption layer with TLS/SIPS at port 5061 at the bottom. Arrows show the flow from device identity verification through secure key negotiation to encrypted audio/video content and secure session setup.

Security Layer Protocol Protects
Signaling Encryption TLS (SIPS on port 5061) Call setup, session details
Media Encryption SRTP Audio/video content
Key Exchange zRTP, DTLS-SRTP Secure key negotiation
Authentication Digest auth, certificates Caller/device identity
Security Best Practice for IoT Doorbells

Many consumer IoT devices have been compromised due to unencrypted streams. Always ensure:

  1. SRTP enabled - Not just RTP (encrypted vs plaintext audio/video)
  2. TLS for SIP - Port 5061, not 5060
  3. Strong authentication - Not default credentials
  4. End-to-end encryption - zRTP prevents cloud provider access to content

44.11 RTP Packet Structure

Understanding RTP helps diagnose audio/video issues in IoT deployments:

44.11.1 RTP Header Format

RTP packet header structure showing a 12-byte fixed header with Version (2 bits), Padding flag (1 bit), Extension flag (1 bit), CSRC Count (4 bits), Marker (1 bit), Payload Type (7 bits), Sequence Number (16 bits), Timestamp (32 bits), and SSRC (32 bits), followed by optional CSRC identifiers and the payload data.

Field Bits Purpose
V 2 Version (always 2)
P 1 Padding flag
X 1 Extension header present
CC 4 CSRC count
M 1 Marker (frame boundary)
PT 7 Payload type (codec identifier)
Sequence 16 Packet ordering/loss detection
Timestamp 32 Media timing (audio sample count)
SSRC 32 Stream identifier

RTP Header Overhead: 12 bytes minimum (vs CoAP’s 4 bytes, MQTT’s 2 bytes)

The larger header is justified for streaming media because: - Sequence numbers detect packet loss and reordering - Timestamps enable jitter buffering and synchronization - SSRC allows multiple streams in one session

Let’s calculate the latency budget for a smart doorbell and understand why the 150ms threshold matters.

Human perception threshold (ITU-T G.114): Interactive speech requires \(L < 150\) ms end-to-end for natural conversation.

Doorbell latency breakdown: \[L_{\text{total}} = L_{\text{camera}} + L_{\text{encode}} + L_{\text{network}} + L_{\text{decode}} + L_{\text{display}}\]

Component latencies:

  • Camera capture: \(L_c = 33\) ms (30 fps frame time)
  • H.264 encode: \(L_e = 20\) ms (hardware encoder)
  • Network (Wi-Fi + Internet): \(L_n = 50\) ms (RTT/2)
  • Client decode: \(L_d = 15\) ms (phone hardware)
  • Display render: \(L_r = 16\) ms (60 Hz display)

Total: \[L_{\text{total}} = 33 + 20 + 50 + 15 + 16 = 134 \text{ ms}\]

With \(134\) ms \(< 150\) ms, two-way conversation is feasible using RTP/SIP. If we used MQTT (TCP handshake \(\approx 1.5 \times \text{RTT}\)), network latency would be \(75\) ms, pushing total to \(159\) ms, exceeding the threshold and creating awkward conversation delays.

Common Pitfalls

Real-time means meeting a deadline consistently, not necessarily having the lowest average latency. A system with 50ms latency that always meets its deadline is more ‘real-time’ than one with 10ms average but occasional 500ms spikes.

MQTT does not provide timing guarantees — broker processing delays and network jitter make it unsuitable for hard real-time control (motor control, safety systems). Use deterministic protocols like EtherCAT or TSN for hard real-time IoT.

Real-time application protocols perform differently under low vs high message rates. Test protocol latency and jitter at maximum expected message throughput, not just at low load.

:

44.13 Worked Example: Smart Doorbell — Protocol Stack Design with Latency Budget

Scenario: RingView, a video doorbell manufacturer, designs the communication stack for a new 1080p doorbell with two-way audio. When a visitor presses the button, the homeowner’s phone (anywhere on the internet) must show live video and enable conversation within 2 seconds of the button press.

44.13.1 Latency Budget Breakdown

The 2-second end-to-end target must be allocated across every component:

Component Budget Protocol Notes
Button press detection 20 ms GPIO interrupt Hardware latency
Push notification to phone 800 ms MQTT QoS 1 over TCP/TLS Firebase/APNs cloud routing
Phone app wake + UI render 400 ms Local iOS/Android cold start
SIP session setup 200 ms SIP over TLS (port 5061) INVITE → 200 OK → ACK
ICE/STUN NAT traversal 300 ms WebRTC ICE Worst case with TURN relay
First video frame render 280 ms RTP/H.264 over UDP IDR frame + decode
Total 2,000 ms Meets 2-second target

44.13.2 Protocol Selection: Why Not Just MQTT?

RingView’s first prototype used MQTT for everything — including video streaming. Here is why that failed:

Metric MQTT (TCP) for Video RTP (UDP) for Video
Transport TCP (reliable, ordered) UDP (best-effort)
Lost packet handling TCP retransmits — video freezes for 50-200 ms waiting RTP skips — next frame arrives in 33 ms (30 fps)
Jitter buffer Not designed for media timing 60 ms adaptive jitter buffer smooths network variation
Measured latency (p95) 380 ms (TCP retransmissions on cellular) 85 ms (UDP with FEC)
Video freeze events/hour 12-18 (each TCP retransmission = freeze) 0-1 (occasional artifact, no freeze)
Audio round-trip (2-way) 420 ms (above 150 ms threshold — conversation awkward) 110 ms (natural conversation)

44.13.3 Bandwidth and Power Analysis

The doorbell runs on a rechargeable 6,800 mAh battery with a 5V solar trickle charger:

Stream Codec Bitrate Daily Usage Daily Energy
Video (1080p, 30 fps) H.264 High Profile 2.5 Mbps 45 min/day (avg 15 events x 3 min) 112 mAh
Audio (two-way) Opus, 16 kHz 32 kbps (up) + 32 kbps (down) 20 min/day 8 mAh
MQTT (notifications + telemetry) N/A 0.5 kbps average 24 hr keep-alive 15 mAh
Idle (PIR wake, Wi-Fi sleep) N/A 0 23 hr 48 mAh
Daily total 183 mAh
Battery life 6,800 / 183 = 37 days (without solar)

44.13.4 Security Stack

Every real-time stream must be encrypted — a doorbell transmitting unencrypted video is a surveillance vulnerability:

Layer Protocol Key Exchange Cipher
Signaling SIP over TLS 1.3 (port 5061) ECDHE P-256 AES-128-GCM
Media (video + audio) SRTP (not RTP) DTLS-SRTP key derivation AES-128-CTR with HMAC-SHA1
Notifications MQTT over TLS 1.3 (port 8883) Certificate pinning AES-256-GCM
The Hybrid Architecture Lesson

RingView’s production doorbell uses three protocols simultaneously:

  1. MQTT (TCP): Push notifications (“Someone is at your door”), device telemetry (battery level, Wi-Fi RSSI), configuration commands. TCP’s reliability is essential — a missed notification means a missed visitor.

  2. SIP/SRTP (UDP): Live video and two-way audio. UDP’s “skip lost packets” behavior is essential — a 33 ms video frame arriving 200 ms late is worse than not arriving at all (it causes stutter). The next frame is already more relevant.

  3. HTTPS (TCP): Recorded clip upload to cloud storage (30-second pre-roll + event). Reliability matters for recordings — every frame must arrive. TCP’s retransmission overhead is acceptable because recording upload happens in the background after the live session.

The mistake is choosing one protocol for everything. MQTT cannot deliver acceptable live video. RTP cannot guarantee notification delivery. Each protocol serves the task it was designed for.

44.14 Concept Relationships

Understanding how real-time protocols connect to the broader IoT ecosystem helps contextualize their unique role:

Protocol Stack Dependencies:

  • Transport Layer: RTP runs over UDP (not TCP), while SIP can use TCP or UDP for signaling. See Transport Fundamentals for UDP vs TCP characteristics.
  • Application Layer: Real-time protocols are application-layer protocols, but with fundamentally different design goals than REST/HTTP. See Application Protocols Overview for the full landscape.

Architectural Integration:

  • Edge Computing: Real-time stream processing often happens at the edge to meet latency requirements. See Edge-Fog Computing for deployment patterns.
  • IoT Reference Models: Real-time protocols typically operate in the communication and data ingestion layers. See IoT Reference Models for architectural context.

Security Integration:

  • Encryption: SRTP provides media encryption while TLS/DTLS secures signaling. See Cryptography Fundamentals for encryption principles.
  • Authentication: SIP digest authentication and certificate-based auth protect session establishment. See Authentication & Access.

Technology Trade-offs:

  • vs MQTT/CoAP: These messaging protocols optimize for reliability and power efficiency, while RTP/SIP optimize for latency and continuity. Neither is “better”—they solve different problems.
  • vs WebRTC: WebRTC bundles signaling, NAT traversal, and media in one framework; SIP+RTP separates concerns for flexibility.

Complementary Technologies:

  • Codecs: RTP carries encoded media (H.264, H.265, Opus, PCM). Codec choice dramatically affects bandwidth and quality.
  • NAT Traversal: STUN/TURN servers enable WebRTC connections across firewalls and symmetric NATs—critical for IoT devices behind home routers.

44.15 See Also

Protocol Deep Dives:

Network Foundations:

Architecture Patterns:

Security Considerations:

Industrial Applications:

  • OPC-UA - Industrial real-time data exchange (different use case)
  • WirelessHART - Industrial wireless with timing guarantees

Research and Standards:

  • RFC 3550 (RTP/RTCP) - Core real-time transport protocol specification
  • RFC 3261 (SIP) - Session Initiation Protocol specification
  • RFC 3711 (SRTP) - Secure Real-time Transport Protocol
  • WebRTC Working Group - IETF standards for browser-based real-time communication

44.16 Summary

This chapter covered real-time protocols for IoT audio and video applications - the protocols that power video doorbells, baby monitors, smart intercoms, and voice assistants:

44.16.1 Key Concepts Covered

  1. The Fundamental Trade-off: Real-time protocols prioritize latency over reliability. They run over UDP, skip lost packets rather than retransmit, and target <150ms end-to-end delay for acceptable human interaction.

  2. Protocol Stack for Real-Time IoT:

    • SIP (Session Initiation Protocol) - Session control (call setup/teardown)
    • RTP (Real-time Transport Protocol) - Media stream delivery over UDP
    • SRTP - Encrypted media for privacy (ALWAYS use for internet-facing devices)
    • WebRTC - Browser-native real-time communication
  3. When to Use Real-Time vs Messaging Protocols:

    Use RTP/SIP/WebRTC Use MQTT/CoAP
    Continuous audio/video streams Discrete sensor readings
    Two-way voice/video calls Commands and telemetry
    Latency-critical (<150ms) Latency-tolerant (seconds OK)
    High bandwidth (64kbps-2Mbps) Low bandwidth (bytes-KB)
  4. Security Requirements:

    • Always use SRTP (not plain RTP) for media encryption
    • Use TLS/port 5061 for SIP signaling (not port 5060)
    • Implement zRTP for end-to-end encryption
    • Never use default credentials
  5. Hybrid Architectures Are Common: Smart doorbells use MQTT for notifications AND RTP for live streaming - the best protocol for each task.

44.16.2 The 150ms Rule

Human conversation becomes awkward when one-way end-to-end delay exceeds 150 milliseconds (ITU-T G.114). This is why real-time IoT devices use UDP-based protocols (no TCP handshake overhead) and specialized media protocols designed for timing, not guaranteed delivery.

44.17 What’s Next?

Chapter Focus Why Read It
Application Protocols Overview Full landscape of IoT application protocols Understand where RTP/SIP fits among MQTT, CoAP, HTTP, and AMQP in a complete protocol taxonomy
MQTT Fundamentals Publish-subscribe messaging for sensor data Compare guaranteed-delivery MQTT with the latency-first RTP approach used in this chapter
CoAP Fundamentals Lightweight RESTful protocol over UDP Explore the other major UDP-based IoT protocol, optimised for constrained devices rather than streaming
Transport Layer Protocols UDP vs TCP characteristics in depth Build the network-layer understanding needed to diagnose RTP jitter and packet-loss problems
IoT Security Fundamentals Threat models for connected devices Apply security thinking to video doorbells and baby monitors after learning SRTP/zRTP in this chapter
Edge-Fog Computing Processing at the network edge Learn how to place media transcoding and jitter-buffer logic at the edge to reduce end-to-end latency