Voice over IP

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Voice over IP

• Why
• Challenges/solutions
• Voice codec and packet delay

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• Motivation
• Benefits
• Reduce backbone network costs managing a single
  packet backbone instead of multiple backbones
  (packet switching for IP and circuit switching
  for voice).
• No way for TDM networks to support IP traffic
• Reduce access network costs
• Bandwidth saving
• one access line for all services
• Reduce premise network (local area network)
  costs
• Use one network to do everything.

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• Challenges
• Bandwidth management to support carrier grade
  phone calls really need working IP QoS
  mechanism.
• Signaling
• Functionality in telephone system is now very
  complicated. Everything must be re-engineered in
  the corresponding signaling system in IP network.
  SIP and H.323
• Media transport
• Need a protocol to transport the contents. Real
  Time Protocol (RTP).
• Interoperability work with the POTS.

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• VoIP and QoS
• Major challenges delay and delay
  variation(Jitter).
• Voice applications are usually interactive.
• delay requirement for a telephone system
  150ms-250ms.
• The sources of delay in a voice over IP system
• OS delay 10s-100s milliseconds
• Voice processing delay DSP 10s milliseconds,
  Sound cards 20-100 milliseconds.
• Look-ahead processing delay coding may need to
  know the next few samples (5ms-7.5ms).
• Packetization delay for voice samples multiple
  sample are usually packed into a packet to save
  bandwidth.
• (n-1)0.125us 40 0.125 50ms
• Packetization delay for voice packet (n-1)t, can
  be quite large.
• Modem delay 20-40ms per modem.

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• The sources of delay in a voice over IP system
  (continue)
• Ingress/egress delay transmission delay at the
  access line. 50 bytes on a 33Kbps access line 50
  8 / 33 12 ms
• Network delay 15ms propagation delay for 3000km
  wires. 100ms all together.
• Total delay
• Gateway to gateway roughly 180ms (100ms network
  delay).
• Desktop to desktop roughly 450ms.
• Delay control mechanism network priority
  mechanisms, end hosts priority mechanism, edge
  equipment design (IP QoS Real time Operating
  Systems voice hardware)

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• Source jitter
• Network network conditions vary at different
  times.
• Non-real time OS samples processed at different
  time.
• Jitter control buffering at the destination.
• QoS parameters
• Accuracy
• Latency
• Jitter
• Codec quality
• QoS control mechanisms sender-based,
  network-based and receiver-base

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• Sender-based
• Retransmissions
• Forward error correction
• Interleaving
• Receiver-based
• Switching to lower bandwidth encoding
• Concealment (silence insertion, noise insertion,
  repeat previous packet, repeat and fade,
  interpolate).
• Network-based IP QoS

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• Voice codes/packet delay and RSVP
• Codec kbps sample size(bits) no. of
  samples no. of bytes delay
• G.711 64 8
  80 80
  10ms
• G.722 64 8
  160 160
  20ms
• G.726 16(24) 2(3/4/5)
  80 20 10ms
• G.726 16 2
  240 60
  30ms
• Issues in Media transfer
• RTP/UDP/IP/link layer protocol
• Protocol overheads 12 bytes RTP header, 8 bytes
  UDP header, 20 bytes IP header.
• G.726 16kbps encoding 20 bytes payload. 33
  link efficiency.

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• Mapping voice stream into TSpec in RSVP
• G.726 16kbps encoding with a packet time of 10 ms
• TSpec Bucket depth, b
• Bucket rate r
• Peak rate p
• minimum policed unit m
• Maximum packet size M
• How to map?

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• Reducing header overheads
• Frame packing
• More frames in one packet
• Less overhead
• Less number of total packets in the system
• Problem?
• RTP multiplexing
• Put multiple frames from different calls in one
  packet
• RTP header compression
• Most fields in the headers are fixed throughout a
  session.
• Record a context id in each router and use the id
  to decide what to do. Reduce RTP/UDP/IP headers
  to 10 bytes.
• Need path setup
• No longer native IP packets.