---
title: "VoIP and IP interconnection; VoIP operation"
subject: "Internet and Intranet"
chapter: 7
key: "semester_8/internet_and_intranet:voip-ip-interconnection-operation"
source: https://www.bctengineer.com/q/semester_8/internet_and_intranet/voip-ip-interconnection-operation
site: BCT Engineering
---

# VoIP and IP interconnection; VoIP operation

Internet and Intranet · Chapter 7: Internet and Intranet Applications

- Asked in: 2073 Bhadra, 8 marks
- 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/internet_and_intranet/voip-ip-interconnection-operation
- Whole chapter: https://www.bctengineer.com/md/semester_8/internet_and_intranet/chapter-7

## Question

What is VoIP and IP Interconnection? Discuss basic operation of VoIP.

## Short answer

VoIP works by signalling (SIP/H.323) for call setup, then digitizing, compressing, packetising voice into RTP/UDP/IP packets, transmitting across the IP network, de-jittering and decoding at the receiver. A gateway converts VoIP to and from PSTN; requirements include low delay, low jitter and under 1% loss.

## Answer

#### VoIP

**VoIP** carries voice as data packets over an IP network. It lets calls be made over the internet or an intranet at low cost, and between computers, IP phones and the ordinary telephone network.

#### IP interconnection

**IP interconnection** is the linking of IP networks of different operators, ISPs and carriers so that traffic and calls pass between them. It is done by peering or transit at an IXP, under agreed technical and commercial terms.

#### Basic operation of VoIP

1. **Call set-up (signalling).** The caller dials. A signalling protocol (SIP or H.323) finds the called party's address (via a SIP proxy/registrar or gatekeeper), checks availability and negotiates codecs. The phone rings.
2. **Digitisation.** The microphone's analogue voice is sampled (8,000 samples/second) and converted to digital (ADC), for example 64 kbps by G.711.
3. **Compression (codec).** A codec such as G.729 (8 kbps) or G.723 reduces bandwidth. Silence suppression saves more.
4. **Packetisation.** Digital voice is cut into small frames (10-30 ms) and wrapped with RTP, UDP and IP headers.
5. **Transmission.** Packets are routed across the IP network. RTP gives sequence numbers and time stamps, and RTCP gives quality feedback. QoS marking keeps delay low.
6. **Reception and de-jitter.** The receiver collects packets in a jitter buffer, reorders them and hides lost packets.
7. **Decoding.** Data is decompressed and converted back to analogue (DAC) for the speaker.
8. **Call release.** On hang-up, signalling ends the session.

```text
 IP Phone/PC --> [ADC+codec] --> [RTP/UDP/IP] --> IP network
                                                      |
 Speaker <-- [DAC+decode] <-- [jitter buffer] <------+
        Signalling (SIP/H.323) sets up and ends the call
```

**Calls to the PSTN** pass through a **VoIP gateway** that converts IP packets to telephone circuits and signalling (SS7). A **softswitch/IP-PBX** manages calls and billing.

**Requirements:** one-way delay below 150 ms, jitter low, packet loss under 1%, and a bandwidth of about 30-90 kbps per call including headers.

## Check yourself

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

1. What is the role of SIP or H.323 in a VoIP call?
   A. They carry the voice samples
   B. They set up the call, locate the called party, negotiate codecs and end the session
   C. They compress the speech
   D. They provide DNS
   Answer: B. Signalling handles call set-up and release, while RTP carries the voice. (Key point: Signalling)

2. How is digital voice sent over the network?
   A. Cut into 10-30 ms frames and wrapped with RTP, UDP and IP headers
   B. Sent as one long file
   C. Sent over TCP only
   D. Sent without headers
   Answer: A. Small frames with RTP/UDP/IP headers are transmitted; RTP supplies sequence numbers and timestamps. (Key point: Packetisation)

3. What is the purpose of the jitter buffer at the receiver?
   A. To collect, reorder and smooth packets and hide lost ones before decoding
   B. To encrypt the voice
   C. To compress the voice more
   D. To set up the call
   Answer: A. Packets arrive with varying delay; the buffer restores order and timing. (Key point: Jitter buffer)

4. What does a VoIP gateway do for calls to the PSTN?
   A. Converts IP packets to telephone circuits and signalling (SS7)
   B. Only measures the bandwidth
   C. Provides the IP address
   D. Removes the codec
   Answer: A. The gateway links the IP network with the circuit-switched telephone network. (Key point: PSTN gateway)

## Related in Internet and Intranet

- [VoIP and IP interconnection definitions](https://www.bctengineer.com/question.md?key=semester_8%2Finternet_and_intranet%3Avoip-ip-interconnection-define) — Internet and Intranet, chapter 7
- [UMS features, VoIP in UMS, IP clearing house](https://www.bctengineer.com/question.md?key=semester_8%2Finternet_and_intranet%3Aums-voip-ipch) — Internet and Intranet, chapter 7
- [VoIP and IPI issues in Nepal](https://www.bctengineer.com/question.md?key=semester_8%2Finternet_and_intranet%3Avoip-ipi-issues-nepal) — Internet and Intranet, chapter 7