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IP and soft switching for PC-to-PSTN calls

Short answer

Skype sends compressed voice packets over IP to a gateway, which converts them to SIP/RTP. A softswitch controls call setup, routing, and billing, and directs a media gateway to convert packets to 64 kbps PCM. SS7 signalling connects the call to the PSTN exchange.

Explain how IP switching and soft switching allow a call to be made from a personal computer with Internet access and running Skype to a telephone connected to PSTN?

A call from a PC running Skype to an ordinary telephone crosses two different networks: the packet-switched Internet (IP switching) and the circuit-switched PSTN. A softswitch and gateways join them. (Skype itself uses its own proprietary protocol; standard SIP/RTP is used from Skype's gateway onwards, inside the operator network.)

PC (Skype client)
   | voice + signalling in Skype's own
   | (proprietary) protocol over UDP/IP
   v
Internet routers (IP switching)
   |
   v
Skype "SkypeOut" gateway --> SIP + RTP
   |
   v
+---------------+  H.248  +------------------+
| Media Gateway |<------->|    Softswitch    |
| pkt <-> TDM   |         | call control,    |
+---------------+         | routing, billing |
   | E1 / PCM             +------------------+
   v                               | SIGTRAN
PSTN exchange <--SS7 ISUP-- Signalling Gateway
   |
   v
Telephone

Role of IP switching

  • The Skype client digitises and compresses the voice (e.g. SILK, or G.729 for calls to phones) and puts about 20 ms of speech in each packet, sent over UDP/IP using Skype's own protocol.
  • Internet routers switch these packets hop by hop using the destination IP address, with no fixed circuit. The packets reach Skype's "SkypeOut" gateway, which passes the call as standard SIP signalling and RTP voice to a partner telecom operator.

Role of the softswitch

A softswitch is call-control software running on servers. It does the job of the control part of a classic exchange, separate from the media path.

  • Receives the call request (dialled number) from Skype's gateway as a SIP INVITE.
  • Authenticates the user and checks credit (billing).
  • Chooses the route and the gateway nearest the called number.
  • Controls the media gateway using MGCP/H.248 (Megaco).
  • Maps SIP signalling into SS7 ISUP messages (IAM, ACM, ANM, REL), which the signalling gateway delivers to the PSTN.

Call steps

  1. The user dials +977-1-xxxxxxx in Skype; the request goes over IP to Skype's servers, which forward it as SIP to the operator's softswitch.
  2. The softswitch verifies the account and selects a gateway and trunk.
  3. The signalling gateway sends an ISUP IAM to the PSTN exchange; the exchange rings the phone and returns ACM.
  4. When the phone is answered, ANM comes back; the softswitch tells the media gateway to connect the RTP stream to a TDM time slot.
  5. The media gateway converts packets to 64 kbps PCM (G.711) and back, adding jitter buffering and echo cancellation.
  6. At hang-up, BYE/REL messages release the slot, and the softswitch records the call for billing.

So IP switching carries the voice packets across the Internet, while the softswitch provides the intelligence (signalling, routing, billing) and controls the gateways that join the Internet to the PSTN.

✓ Answer checked by Claude Opus 5.5 (high effort)

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