---
title: "IP telephony / Voice over IP"
subject: "Telecommunications"
chapter: 8
key: "semester_8/telecommunications:ip-telephony-voip"
source: https://www.bctengineer.com/q/semester_8/telecommunications/ip-telephony-voip
site: BCT Engineering
---

# IP telephony / Voice over IP

Telecommunications · Chapter 8: Data Communication

- Asked in: 2079 Chaitra, 4 marks; 2079 Bhadra, 5 marks; 2070 Asar, 12 marks; 2070 Chaitra, 9 marks (asked 4 times)
- Video: [What is VoIP?](https://www.youtube.com/watch?v=n_b2sJP2pBk) — PowerCert Animated Videos
- Answer checked by Claude Opus 5.5 (high effort) on 2026-10-14
- Page: https://www.bctengineer.com/q/semester_8/telecommunications/ip-telephony-voip
- Whole chapter: https://www.bctengineer.com/md/semester_8/telecommunications/chapter-8

## Question

Briefly explain IP telephone system (IP telephony or Voice over IP).

## Short answer

IP telephony (VoIP) transmits voice as data packets over IP networks. Voice is digitized, compressed, packetized with RTP/UDP/IP, and reassembled at the receiver using a jitter buffer. Signalling uses SIP or H.323; components include IP phones, gateways, and softswitches.

## Answer

**IP telephony (Voice over IP, VoIP)** is the transmission of voice calls as data packets over an IP network (internet or private IP network) instead of over dedicated circuits of the PSTN. Voice is digitised, compressed, put into IP packets and sent; at the far end the packets are reassembled and played out.

#### Basic operation

```text
 Speech -> A/D + codec -> packetise (RTP/UDP/IP)
                              |
                         IP network
                              |
 Speech <- D/A + decoder <- jitter buffer <- depacketise
```

1. **Digitisation:** the analog voice is sampled at 8 kHz and coded.
2. **Compression (codec):** G.711 (64 kbps), G.729 (8 kbps), G.723.1 (5.3/6.3 kbps), Opus.
3. **Packetisation:** e.g. 20 ms of speech per packet, with RTP, UDP and IP headers.
4. **Transmission** over the IP network; packets may take different paths.
5. **Jitter buffer** at the receiver smooths variable delay.
6. **Decoding and playout**; lost packets are concealed.

**Signalling** sets up and ends calls: **SIP** (IETF) or **H.323** (ITU-T), plus MGCP/H.248 for gateways. **RTP/RTCP** carry voice and quality reports.

#### Components

- **IP phones and softphones** (terminals).
- **Gateway:** converts between PSTN (TDM) and IP for calls to normal phones.
- **Gatekeeper (H.323) / SIP proxy and registrar:** address translation, admission control, routing.
- **IP-PBX / softswitch:** call control for an enterprise or carrier.
- **MCU:** multipoint conferencing.

#### Call configurations

| Type | Path |
|---|---|
| PC to PC | both ends on the internet (e.g. app calls) |
| PC to phone | IP to a gateway, then PSTN |
| Phone to phone | PSTN – gateway – IP – gateway – PSTN |
| IP phone to IP phone | enterprise IP-PBX |

#### Quality of service issues

- **Delay:** one-way delay should be below about 150 ms (ITU-T G.114) for good conversation.
- **Jitter:** variation in packet delay; handled by jitter buffers.
- **Packet loss:** causes gaps; handled by loss concealment, FEC.
- **Echo:** needs echo cancellers.
- QoS techniques: DiffServ priority marking, MPLS, bandwidth reservation.

#### Advantages

- Much lower cost, especially for long distance and international calls.
- One network for voice, video and data (convergence).
- Better bandwidth use through compression and silence suppression.
- Rich features: video, presence, conferencing, mobility of number.

#### Disadvantages

- Quality depends on network; delay and jitter can be noticeable.
- Needs power and internet; emergency calls and location are harder.
- Security threats: eavesdropping, spam over IP telephony, denial of service.

VoIP is now the basis of enterprise telephony, operator NGN/IMS cores and VoLTE in 4G mobile networks.

## Check yourself

The answer's key points as multiple-choice questions, with the correct option and why.

1. How does VoIP carry a call?
   A. On a dedicated circuit of the PSTN
   B. Voice is digitised, compressed and sent as IP packets reassembled at the far end
   C. As analog signals over IP
   D. Only through satellites
   Answer: B. Voice is digitised, compressed, packetised and sent over an IP network. (Key point: VoIP principle)

2. What is the jitter buffer for?
   A. To compress voice
   B. To encrypt packets
   C. To cancel echo
   D. To smooth variable packet delay at the receiver
   Answer: D. Jitter is variation in packet delay; the receiver's jitter buffer smooths it. (Key point: Jitter buffer)

3. What one-way delay does ITU-T G.114 give for good conversation?
   A. Below about 150 ms
   B. Below about 15 ms
   C. Below about 1.5 s
   D. Below about 500 ms
   Answer: A. One-way delay should be under about 150 ms. (Key point: Delay limit)

4. Which codec/rate pair is correct?
   A. G.711 at 8 kbps
   B. G.729 at 64 kbps
   C. G.729 at 8 kbps
   D. G.723.1 at 64 kbps
   Answer: C. G.711 is 64 kbps, G.729 is 8 kbps, G.723.1 is 5.3/6.3 kbps. (Key point: Codec bit rates)

## Related in Telecommunications

- [IP and soft switching for PC-to-PSTN calls](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Aip-soft-switching-skype-pstn) — Telecommunications, chapter 8
- [Network planning for remote area telephony](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Anetwork-planning-remote-area) — Telecommunications, chapter 1
- [Telephone exchange and switching hierarchy](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Aexchange-switching-hierarchy) — Telecommunications, chapter 1, asked 3×
- [Switching hierarchy and evolution until the telephone](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Aswitching-hierarchy-telephone-evolution) — Telecommunications, chapter 1, asked 3×
- [Telephone exchange structure; TST vs STS](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Aexchange-structure-tst-sts) — Telecommunications, chapter 4
- [Analog vs digital multiplexing; 24-channel system](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Aanalog-digital-mux-24-channel) — Telecommunications, chapter 3

## Related in other subjects

- [FDM hierarchy in telephone system](https://www.bctengineer.com/question.md?key=semester_5%2Fdata_communication%3Afdm-hierarchy-telephone) — Data Communication, chapter 6
- [Numerical: cellular vs non-cellular voice channels](https://www.bctengineer.com/question.md?key=semester_7%2Fwireless_communication%3Acellular-vs-noncellular-channels) — Wireless Communication, chapter 2
- [OFC system; pros and cons over metal wire](https://www.bctengineer.com/question.md?key=semester_6%2Fpropagation_and_antenna%3Aofc-system-vs-metal-wire) — Propagation and Antenna, chapter 6
- [TDM of 24 voice signals: PAM bandwidth and PCM sampling rate](https://www.bctengineer.com/question.md?key=semester_6%2Fcommunication_systems%3Atdm-24-voice-pam-pcm) — Communication Systems, chapter 5