VoIP

1. VoIP Voice over IP

2. Contents • Introduction • How VoIP works • Voice over IP Scenarios • The „Pros and Cons“ • RTF (Real-time Transport Protocol) • H.323 Standard • Speech Samples • Future Aspects

3. Introduction What is VoIP ? • VoIP = Voice over Internet Protocol • Transmission of voice and video over data-networks (internet, intranet), using the Internet protocol (IP) • Possible variants using VoIP : • PC to PC • PC to Phone and the other way around • Phone to Phone using the Internet / Intranet

4. How VoIP works The Classical Approach:Analog Transmission Signal is transferred by a wire microphone Microphone transforms acoustic waves into "current fluctuations" Loudspeaker Loudspeaker transforms the signal back into sounds

5. How VoIP works The Classical Approach :Digital Transmission Usual: 256 different values (256=2^8, => 8 bit); 8000 measures per second => 64 kBit/s 01110001 11000001 ……… Microphone transforms acoustic waves into "current fluctuations" In time intervals the amperage is measured and its value is transferred Converted back into current fluctuations and acoustic waves transmission of 0 and 1 as on and turn-off processes; nearly error free

6. How VoIP works Tomorrow (?):Voice over IP Transmission of media (audio,video) over the Internet Protocol: The Internet works package-oriented: data stream is distributed on packets, which are sent independently to the target • This means for the media (speech), like you would: • record it on a tape • cut the tape into pieces • put the pieces in envelopes • at the destination paste the pieces in the correct • order and play the tape • … and this all in real-time (almost)!

7. How VoIP works How VoIP works • General approach in case of submitting speech: • Recording and digitalisazion of speech • Segmentation of data packets • The packets are transported over the internet to the receiver • The receiver‘s hardware turns the data back into speech • Timestamp guarantees correct time and order

8. How VoIP works Synchronisation through Timestamp sampled Sampling Instants 56 58 59 75 76 77 78 79 95 96 57 97 98 99 packetized 58 78 98 Timestamp sent 58 78 98 118 138 158 received 58 78 98 118 138 158 replayed 58 78 98 118 138 158

9. Site A Site B Internet IPRouter VoiceSwitch VoiceSwitch IPRouter PC PC Intranet Modem Modem Private VoiceNetwork Phone Phone PSTN Fax Fax Voice over IP Scenarios The Classical Approach:Separate Voice and Data Networks

10. Site A Site B PC withVoice/Fax PC withVoice/Fax IPRouter IPRouter Internet Intranet IP Phone IP Phone PSTN Gateway PSTN Gateway PSTN Voice over IP Scenarios The Future Approach:Voice/Fax over IP - A Unified Network

11. Site A Site B PC withVoice/Fax PC withVoice/Fax IPRouter IPRouter VoiceSwitch VoiceSwitch Internet Intranet IP Phone PSTN Gateway PSTN Gateway IP Phone PSTN Phone Phone Fax Fax Voice over IP Scenarios The Intermediate Approach:Voice/Fax over IP - A Unified Network

12. Site A Site B Internet Intranet Voice over IP Scenarios The Gamers Approach:Use of “Roger Wilco” PC with Headset Roger Wilco Server PC with Headset Roger Wilco Client

13. The „Pros and Cons“ • Advantages • Cost savings on long distance calls • Less (no) need for private telephone networks • Single RJ-45 connector at the workplace for all services • Enables new multimedia features, e.g. human operator assisted e-commerce • Problems / Open Questions • Control of delay, jitter and packet loss over IP-based networks • QoS guarantees • Bandwidth

14. Contents • Introduction • How VoIP works • Voice over IP Scenarios • The „Pros and Cons“ • RTF (Real-time Transport Protocol) • H.323 Standard • Speech Samples • Future Aspects

15. Real-time Transport Protocol (RTP) Real-time Transport Protocol (RTP) • RTP version 2 is specified by RFC 1889 • RTP covers functions such as • Payload type identification (which codec and Framing) • arranging the packages by sequence numbers synchronisation by time stamps (playing time of the individual Samples or Frames; • Synchronisation of several Media Streams • quality control and statistics • RTP is defined independent of transportation protocol, sets however typically on UDP/IP • RTP contains no mechanisms to the safety device of the quality of service (QoS)

16. Real-time Transport Protocol (RTP) RTP and RTCP • The concept covers two closely linked protocols • RTP (real-time transport Protocol): transport the Media Stream • RTCP (real-time transport control Protocol): informed about the participants attached at the Media Stream and the quality of service (QoS) • For each Media Stream and each direction, received from them, a separate RTP session opens, you get • an IP address (multicast address for Conferencing) • a pair of UDP ports • n for the Media Stream (default 5004) • n+1 for RTCP (default 5005)

17. Real-time Transport Protocol (RTP) RTCP: Paket-Typs and scalability • RTCP reports are generated by all transmitters and receivers of the session in regularly intervals (statistic information) • the interval must be selected in such a way that the total load remains within limits (recommended: approx.. 5% of the range of the session) • very large groups to make possible (with thousands of participants), the interval computed due to the • amount of active participant • the extent of the individual report

18. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|X| CC |M| PT | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | timestamp | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | contributing source (CSRC) identifiers | | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | optional header extension | | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Real-time Transport Protocol (RTP) Header Format

19. IP (20) UDP (8) RTP (12) Payload (20) 24 kbit/s compressed RTP (2 ... 4) cRTP Payload(20) ~10 kbit/s Real-time Transport Protocol (RTP) Overhead und Header Compression • RTP header Overhead: Example • compressed language - 8 kbps • every 20 ms a RTP package with 20 oktetts Payload • 40 oktetts header per package - 24 kbps • RTP header compression: • reduces Overhead • router decompression necessarily - > high processor load • Slow i-net connection (e.g. VoIP over V.34 modem)

20. Real-time Transport Protocol (RTP) VoIP Delay Budget (example: G.711) *) depends on uses codec • Coding and framing 20 ms *) • Packetization 20 ms • Output queuing 0 ... 8 ms • Access (up) link transmission • Backbone network transmission t • Access (down) link transmission • Input queuing not relevant • Jitter buffer 4 ... 40 ms • Decoding 1 ms *) ITU-T G.114 Recommended (150 ms) Total 45 ... 89 + t [ms]

21. LAN H.323 Standard H.323 : Packet-BasedMultimedia Communications Systems H.225.0 Layer Video Codec H.261, H.263 Video I/O Equipment RTP Audio Codec G.711, G.722 Audio I/O Equipment User Data Applications T.120, etc. System Control RAS Control System Control User Interface Q.931 Call Setup H.323 H.245 Control

22. Excellent 5 Speech Quality Good 4 Fair 3 Poor 2 64 32 16 8 4.8 2.4 Bit Rate [kbps] MOS - Mean Opinion Score 5 - Excellent 4 - Good 3 - Fair 2 - Poor 1 - Bad

23. Speech Samples Speech Samples G. 711 64 kbps LPC 2.4 kbps GSM 13 kbps Single Speaker Music Bit Errors 0.1% Bit Errors 1%

24. Future Aspects Future Aspects

25. Thanks for your attention