Telecommunication Systems and
Wireless Networks
Telecommunication Services and Voice over IP
Telecommunication services: where VoIP fits
Services
Voice calling • Messaging • Video meetings • Contact centers • Emergency services
Control & media protocols
Signaling (SIP / IMS) • Media transport (RTP/RTCP) • Security (TLS/SRTP)
Networks
Access (Wi-Fi, DSL, cable, fiber, 4G/5G) • IP core (routing, QoS, peering) • Legacy interconnect (PSTN, gateways)
VoIP = voice service delivered over IP (not dedicated circuit-switched PSTN)
Circuit switching vs packet switching (voice)
Circuit-switched (PSTN) Packet-switched (IP/VoIP)
• Dedicated end-to-end path (reserved resources) • Voice encoded → packetized → routed like other
• Predictable delay and jitter IP traffic
• Signaling + media tied to the circuit • Statistical multiplexing (shared links)
• Efficient for constant-rate voice; less flexible for • Variable delay/jitter; loss possible (best effort)
data • Needs buffering, QoS, and robust codecs
VoIP endpoints: ATA vs IP phone vs softphone
ATA (Analog Telephone Adapter) IP Phone Softphone
• App on PC/mobile
• Reuse legacy handset • Ethernet/Wi-Fi endpoint
• Uses headset/mic
• Good for home migration • Often supports PoE
• Great for remote work +
• Gateway: analog digital • Enterprise features (PBX/UC)
WebRTC
VoIP split: signaling vs media
Setup → Talk → Teardown
Control plane (Signaling)
• SIP messages: REGISTER, INVITE, 200 OK, ACK, BYE
• Negotiates media parameters via SDP
User plane (Media)
• RTP carries audio/video packets
• RTCP carries feedback: loss, jitter, timing
SIP
Session Initiation Protocol (signaling)
RTP / RTCP
Real-Time Transport Protocol (media) + control feedback
QoS for VoIP: what can go wrong (and what we do)
Key metrics Typical mitigations
• One-way delay (latency): humans notice
conversational lag • Packet prioritization (DiffServ/queuing) on managed
• Jitter: variation in delay → needs jitter buffer networks
• Packet loss: causes gaps/robotic audio; PLC can mask • Adaptive jitter buffer and packet-loss concealment
small loss • Codec choice: bandwidth vs robustness vs CPU
• Echo: worse with higher delays → echo cancellation • RTCP feedback + congestion control (esp. for video)
matters
Rule of thumb: keep one-way delay well below 150 ms for “good” voice; 150–400 ms can be acceptable with care and echo control.
Codecs & packetization: why overhead matters
Common audio codecs (examples) Header overhead (IPv4/UDP/RTP)
• IP header: 20 bytes
• G.711 (PCM): high quality, ~64 kbps payload
• UDP header: 8 bytes
• Opus: wideband, adapts bitrate/packet loss
• RTP header: 12 bytes
• G.729 / AMR: lower bitrate, more compression
• Total: 40 bytes (before link-layer headers)
Packetization interval (typical)
Example (G.711, 20 ms):
~10–30 ms of audio per RTP packet (trade-off: overhead vs Payload: 64 kbps × 0.02 s = 1280 bits = 160
delay) bytes
Headers: ~40 bytes
Overhead ≈ 40 / (160 + 40) = 20%
Security & NAT traversal (practical VoIP)
Security goals Why NAT is tricky
• Protect signaling (who is calling whom?) • SIP/SDP carry IP:port info that NAT may rewrite
• Protect media (what is being said?) • RTP uses dynamic UDP ports
• Prevent spoofing/replay
Typical approach
Common building blocks • STUN: discover public mapping
• SIP over TLS (sips:) • TURN: relay when direct path fails
• SRTP/SRTCP for media confidentiality + integrity • ICE: tries candidate paths and picks the best
Operational note: emergency calling and lawful intercept requirements may affect VoIP deployments.
Recap & quick check
Key takeaways
• VoIP runs voice over IP networks; quality depends on delay/jitter/loss.
• SIP is signaling (setup/modify/terminate); RTP is the media transport.
• RTP adds sequencing/timestamps; RTCP adds feedback and synchronization.
• Real deployments need security (TLS/SRTP) and NAT traversal (ICE).
Quick check (1–2 minutes)
• 1) In SIP, which message confirms the final response to INVITE?
• 2) In RTP, which fields help detect loss and reorder packets?
• 3) Why is TCP usually a bad fit for real-time voice media?