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Voice over IP. Why Challenges/solutions Voice codec and packet delay. 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

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voice over ip
Voice over IP
  • Why
  • Challenges/solutions
  • Voice codec and packet delay
slide2
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.
slide4
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.
slide5
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.
slide6
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)
slide7
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
slide8
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
slide9
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.
slide10
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?

slide11
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.
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