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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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Presentation Transcript
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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.


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