- 2074 Asoj · 8 marks
PC-to-phone calls via internet and PSTN
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.
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.
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
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
- User logs in to the operator's VoIP service from the PC application.
- User dials the phone number; a SIP INVITE / H.225 Setup goes to the softswitch.
- Softswitch checks the account, chooses a gateway and seizes a free E1 channel.
- Gateway sends ISUP IAM to the exchange; the exchange rings the phone and returns ACM; ringback tone is played to the PC.
- When the called party answers (ANM), the speech path opens: PC ↔ Internet ↔ gateway ↔ PSTN ↔ phone.
- 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.
✓ Answer checked by Claude Opus 5.5 (high effort)
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