---
title: "PC-to-phone calls via internet and PSTN"
subject: "Telecommunications"
chapter: 8
key: "semester_8/telecommunications:pc-to-phone-call-internet"
source: https://www.bctengineer.com/q/semester_8/telecommunications/pc-to-phone-call-internet
site: BCT Engineering
---

# PC-to-phone calls via internet and PSTN

Telecommunications · Chapter 8: Data Communication

- Asked in: 2074 Asoj, 8 marks
- Answer checked by Claude Opus 5.5 (high effort) on 2026-10-14
- Page: https://www.bctengineer.com/q/semester_8/telecommunications/pc-to-phone-call-internet
- Whole chapter: https://www.bctengineer.com/md/semester_8/telecommunications/chapter-8

## Question

Nepal Telecom offers a service that allows you to make a phone call from a PC to an ordinary phone. This means that voice call must pass through both the internet and through a telephone network. Discuss how this might be done.

## Short answer

Voice from the PC is packetized as RTP/UDP/IP and sent over the Internet to a VoIP gateway. The gateway converts between IP packets and 64 kbps PCM for the PSTN, while a softswitch/gatekeeper translates SIP/H.323 call signalling to SS7 ISUP and handles authentication, routing, and billing. Echo cancellation, jitter buffering, and codec transcoding are done in the gateway.

## Answer

A PC-to-phone call is a **VoIP (Voice over IP)** call in which one end is a packet network (the Internet) and the other end is the circuit-switched PSTN. The telecom operator joins the two with a **VoIP gateway** controlled by a **gatekeeper/softswitch**.

#### System arrangement
```text
PC + softphone (mic, speaker)
   | voice in RTP/UDP/IP; SIP or H.323 signalling
   v
Internet (ISP routers)
   v
+------------------- Operator (e.g. NT) -----------+
| Gatekeeper/Softswitch: login, number translation,|
|   routing, billing (prepaid account / PIN)       |
|              | H.248 / MGCP control              |
| VoIP Gateway: packets <-> 64 kbps PCM (E1 TDM)   |
+--------------+-----------------------------------+
               | E1 trunk + SS7 ISUP signalling
               v
         PSTN local/tandem exchange --> called phone
```

#### Functions of each part
- **PC softphone:** samples the voice, encodes it with a codec (G.711 64 kbps, G.729 8 kbps, G.723.1 6.3 kbps), places 20–30 ms of speech in each RTP packet, adds UDP and IP headers. It also does silence suppression and has a jitter buffer for received packets.
- **Internet:** routes the packets like any other IP traffic (best-effort, no reserved circuit), so delay, jitter and loss vary.
- **Gatekeeper / softswitch:** registers and authenticates the user (account and password or PIN), checks balance, translates the dialled E.164 telephone number into the correct gateway address, controls the gateway and records call details for billing.
- **VoIP media gateway:** converts packets to PCM samples and places them in an E1 time slot towards the PSTN, and the reverse for the other direction. It does codec transcoding (e.g. G.729 to G.711), echo cancellation, jitter buffering and packet-loss concealment.
- **Signalling gateway:** translates SIP/H.323 call signalling to **SS7 ISUP** (IAM, ACM, ANM, REL) used by the PSTN.

#### Call procedure
1. User logs in to the operator's VoIP service from the PC application.
2. User dials the phone number; a SIP INVITE / H.225 Setup goes to the softswitch.
3. Softswitch checks the account, chooses a gateway and seizes a free E1 channel.
4. Gateway sends ISUP IAM to the exchange; the exchange rings the phone and returns ACM; ringback tone is played to the PC.
5. When the called party answers (ANM), the speech path opens: PC ↔ Internet ↔ gateway ↔ PSTN ↔ phone.
6. On hang-up from either side, BYE/REL messages free the channel; the softswitch charges the account by duration.

#### Quality issues and remedies
| Problem | Remedy |
|---|---|
| Delay (should be < 150 ms one-way) | Short packets, low-delay codecs, nearby gateway |
| Jitter | Jitter buffer at PC and gateway |
| Packet loss | Loss concealment, FEC |
| Echo (hybrid at PSTN side) | Echo canceller in gateway |
| Low bandwidth at PC | Compressed codec (G.729) |

In this way the call is carried as packets over the Internet up to the operator's gateway, and as a normal 64 kbps circuit over the PSTN from the gateway to the telephone.

## Check yourself

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

1. What does the VoIP media gateway do in a PC-to-phone call?
   A. Converts IP packets to 64 kbps PCM in an E1 time slot and vice versa
   B. Only authenticates the user
   C. Only routes e-mail
   D. Generates the voice samples at the PC
   Answer: A. The media gateway joins the packet network and the circuit-switched PSTN by converting packets to PCM and back. (Key point: Role of VoIP gateway)

2. Which task belongs to the gatekeeper/softswitch?
   A. Silence suppression at the PC
   B. Authenticating the user, translating the E.164 number to a gateway, controlling the gateway and billing
   C. Regenerating weak signals
   D. Cancelling echo on the line
   Answer: B. The softswitch handles registration, balance check, number translation, gateway control and call records. (Key point: Role of gatekeeper/softswitch)

3. Which conversion does the signalling gateway perform?
   A. SIP/H.323 to SS7 ISUP
   B. TCP to UDP
   C. G.711 to G.729
   D. RTP to HTTP
   Answer: A. The signalling gateway translates IP call signalling (SIP/H.323) into SS7 ISUP used by the PSTN. (Key point: Signalling conversion)

4. Which is the correct order of events?
   A. ANM, IAM, INVITE, ACM
   B. INVITE to softswitch, IAM to exchange, ACM (ringback), ANM when answered
   C. ACM, INVITE, ANM, IAM
   D. BYE, INVITE, IAM, ANM
   Answer: B. The softswitch gets the INVITE, the gateway sends IAM, the exchange rings and returns ACM, and ANM opens the speech path. (Key point: Call set-up sequence)

## 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
- [Short note: Telephone numbering in PSTN](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Ashort-note-pstn-numbering) — Telecommunications, chapter 7, asked 2×
- [Traffic numerical: 800 calls, 50 lost](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Agos-800-calls-50-lost) — Telecommunications, chapter 6, asked 6×
- [Traffic numerical: 600 calls, 40 lost, 120 s](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Agos-600-calls-120s) — Telecommunications, chapter 6, asked 5×
- [Charging plan in a telephone network](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Acharging-plan-telephone) — Telecommunications, chapter 7
- [Role of NTA in Nepal's telecom sector](https://www.bctengineer.com/question.md?key=semester_8%2Ftelecommunications%3Arole-of-nta) — Telecommunications, chapter 7, asked 3×

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- [Case study: Nepal Electricity Authority / Telecom](https://www.bctengineer.com/question.md?key=semester_7%2Forganization_and_management%3Acase-study-nea-nt) — Organization and Management, chapter 9
- [PC and SP in CALL, RET, PUSH, POP](https://www.bctengineer.com/question.md?key=semester_4%2Fmicroprocessor%3Apc-sp-call-ret-push-pop) — Microprocessor, chapter 2
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