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VoIP and IP interconnection; VoIP operation

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.

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

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.
 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.

✓ Answer checked by Claude Opus 5.5 (high effort)

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