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.