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IP telephony / Voice over IP

Short answer

IP telephony (VoIP) transmits voice as data packets over IP networks. Voice is digitized, compressed, packetized with RTP/UDP/IP, and reassembled at the receiver using a jitter buffer. Signalling uses SIP or H.323; components include IP phones, gateways, and softswitches.

Briefly explain IP telephone system (IP telephony or Voice over IP).

IP telephony (Voice over IP, VoIP) is the transmission of voice calls as data packets over an IP network (internet or private IP network) instead of over dedicated circuits of the PSTN. Voice is digitised, compressed, put into IP packets and sent; at the far end the packets are reassembled and played out.

Basic operation

 Speech -> A/D + codec -> packetise (RTP/UDP/IP)
                              |
                         IP network
                              |
 Speech <- D/A + decoder <- jitter buffer <- depacketise
  1. Digitisation: the analog voice is sampled at 8 kHz and coded.
  2. Compression (codec): G.711 (64 kbps), G.729 (8 kbps), G.723.1 (5.3/6.3 kbps), Opus.
  3. Packetisation: e.g. 20 ms of speech per packet, with RTP, UDP and IP headers.
  4. Transmission over the IP network; packets may take different paths.
  5. Jitter buffer at the receiver smooths variable delay.
  6. Decoding and playout; lost packets are concealed.

Signalling sets up and ends calls: SIP (IETF) or H.323 (ITU-T), plus MGCP/H.248 for gateways. RTP/RTCP carry voice and quality reports.

Components

  • IP phones and softphones (terminals).
  • Gateway: converts between PSTN (TDM) and IP for calls to normal phones.
  • Gatekeeper (H.323) / SIP proxy and registrar: address translation, admission control, routing.
  • IP-PBX / softswitch: call control for an enterprise or carrier.
  • MCU: multipoint conferencing.

Call configurations

TypePath
PC to PCboth ends on the internet (e.g. app calls)
PC to phoneIP to a gateway, then PSTN
Phone to phonePSTN – gateway – IP – gateway – PSTN
IP phone to IP phoneenterprise IP-PBX

Quality of service issues

  • Delay: one-way delay should be below about 150 ms (ITU-T G.114) for good conversation.
  • Jitter: variation in packet delay; handled by jitter buffers.
  • Packet loss: causes gaps; handled by loss concealment, FEC.
  • Echo: needs echo cancellers.
  • QoS techniques: DiffServ priority marking, MPLS, bandwidth reservation.

Advantages

  • Much lower cost, especially for long distance and international calls.
  • One network for voice, video and data (convergence).
  • Better bandwidth use through compression and silence suppression.
  • Rich features: video, presence, conferencing, mobility of number.

Disadvantages

  • Quality depends on network; delay and jitter can be noticeable.
  • Needs power and internet; emergency calls and location are harder.
  • Security threats: eavesdropping, spam over IP telephony, denial of service.

VoIP is now the basis of enterprise telephony, operator NGN/IMS cores and VoLTE in 4G mobile networks.

✓ Answer checked by Claude Opus 5.5 (high effort)

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