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

2. 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.

4. 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.

5. 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.

6. 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)

7. 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

8. 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

9. 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.

10. 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?

11. 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.