IP Telephony (Voice Over IP)

1. IP Telephony(Voice Over IP) Patrick Duff Illustrations from Global Knowledge

2. Why Voice Over IP • Cheaper voice communications • Offers the ability to combine a voice network with a data network • Voice Over IP will offer solutions to efficiently send voice (and video) over pre-existing networks

3. Definitions • IP – Internet Protocol; Determines the format of packets and their addresses as they are sent over the network • RTP – Realtime Transport Protocol; part of voice packet that contains digitized audio • Frame – container of data sent over a network • Packet – smaller container sent over a network

4. Definitions • Multiplexers – combine many communications to be transmitted on one line • Codec – encoder/decoder • PCM – pulse code modulation • LAN – Local Area Network • WAN – Wide Area Network • Payload – information delivered by the packet or frame

5. Existing Networks • Already send out data packets • Maximum Transmission Latency is the maximum amount of bytes allowed in a frame (packet) • Ethernet transmission MTUs are 1500, approximately 15 times larger than Voice MTUs

6. So? • It takes 214 ms for an Ethernet frame to reach it’s destination • A Voice packet has to wait for this frame to finish, exceeding the 150 ms ideal for voice communication, and even the 190 ms delay for slightly delayed voice communication • This delay is unacceptable in voice communication

7. Solutions • Prioritizing traffic (voice before data) • Use Frame Relay Access Device to break down data frames into the same size as voice packets • The FRAD will then alternate voice and data packets according to priority • Upgrading to a T1 • Reducing transmission time from 214 ms to 134 ms

8. Multiplexers • A voice communications requires only a 4Khz range of frequency • Existing networks can handle a lot more than this • Multiplexers put multiple communication channels through one wire

9. Time Division Multiplexers • Time Division converts an analog signal into a digital signal • Time Division Multiplexers put each channel into a specific time slot • Each channel has it’s turn to send it’s signal over the network

10. Codecs for Voice Compression • “G.711 is a specific PCM scheme that samples voice 8,000 times per second and converts the 8-bit samples into a digital stream.” • Other codecs • G.729 • G.723.1 Feeser, Stuart. “Introduction to Voice Over IP”

11. G.711 PCM • The G.711 codec uses an 8000 sample per second clock to convert the 8 bit signal into a digital stream • At each point of amplitude, the measure is increased to improve fidelity • This is called companding

12. Companding Methods • µ-law - used in North America • A-law - used in Europe

13. µ-law • 16 logarithmic divisions • Each contains 16 linearly spaced steps • Only has 255 steps • 0 is not used

14. A-law • 16 logarithmic divisions • Each contains 16 logarithmically spaced steps • Uses all 256 steps

15. Size of Voice Packets • Every byte is not sent out as soon as it is encoded • A packet is created with a 58 byte header • Two main sizes of packets are used • 20 ms • 60 ms

16. 20 millisecond packets • Collects data for 20 ms • 50 packets per second • 160 bytes per packet • Calculating overhead • 58 bytes * 8 bits per byte * 50 packets per second = 23.2 Kbps overhead

17. 20 millisecond packets • Benefits • Less delay than 60 ms sampling • Good for LANs or WANs • Disadvantages • More overhead • 50 packets must be sent per second

18. 60 millisecond packets • Collects data for 60 ms • 16 2/3 packets per second • 480 bytes per packet • Calculating overhead • 58 bytes * 8 bits per byte * 16.7 packets per second = 7.75 Kbps overhead

19. 60 millisecond packets • Benefits • Less overhead • Only 16 2/3 packets need to be sent per second • Good for connecting through modems • Disadvantages • More delay

20. 58 byte header • 6 bytes for the MAC address source • 6 bytes for the MAC address destination • 4 bytes for IP address originating source • 4 bytes for IP address destination • 2 bytes for Transport Address Source • The port address at the point of origin • 2 bytes for Transport Address Destination • The port address at the destination

21. 58 byte header (cont.) • 1 byte for priority of the packet • 2 bytes for the sequence number • Where this packet belongs in the total number of packets • 1 byte for the HOP counter • How many routers this packet went through • 2 bytes for the Payload type • What this packet is carrying

22. 58 byte header (cont.) • 4 bytes for the Timestamp header • 4 bytes for the synchronization source • IP address of the computer that created this packet • 1 byte for the Version • Version number of the items • 3 bytes for the Protocol ID • Rules governing hardware address, network address and transport address

23. 58 byte header (cont.) • 4 bytes for flags • 4 bytes for validity of the payload (checksum) • 4 bytes for length • 4 bytes for frame check sequence • Payload validity checked at each router or ethernet switch

24. Packet Switching • Voice Packets contain • RTP = Real Time Protocol • UDP = User Datagram Protocol • IP = Internet Protocol • MAC = Media Access Control

25. When a packet is sent out • Each part of the header directs the packet to it’s destination

26. Voice Over IP in Action • Convert voice to data • Send out data, start creating new packet • What happens when Net Traffic is heavy and packets arrive out of order or in different intervals than sent out?

27. The Jitter Buffer • Collects the packets • Each packet is time stamped according to the 8000 tick per second clock as to when the packet was created • The jitter buffer will collect all the packets and detect time irregularities between packets with a synchronized 8000 tick per second clock

28. The Jitter Buffer • So packets arrived too fast, too slow, or out of order • Step 1, reorder packets • Step 2, determine slowest arriving packet • Step 3, determine delay of packet • Step 4, delay all other packets the same amount of time

29. Silence Suppression • Most of the time during voice communications, one member is not talking • Instead of sending empty packets, don’t send any packets • After not sending packets, how do we re-synchronize with the jitter buffer?

30. Silence Suppression • 20 ms packet example: • Each packet should arrive every 160 ticks • The packets aren’t out of order, but have skipped some time • The jitter buffer looks at the time stamp and delays the packet according to the amount of time between the last packet’s timestamp and the current packet’s timestamp

31. RTP Mixers • RTP mixers combine multiple incoming signals, then sends them to all intended receivers • Source timing isn’t always synchronized • The RTP mixer decides it’s own time stamping and sends the combined signal with a new time stamp

32. Why RTP Mixers? • A conference call with 4 callers • One voice signal goes to the other 3 • If two people talk at the same time • The mixer mixes the two voices and sends it to the two silent people • The two people talking don’t get the mixed signal, only the signal of the other person talking

33. Translator • If two people on the same Voice Over IP call use two different size packets • 60 ms • 20 ms • The translator on each side will accept the incoming data even though it’s sending out data in a different format

34. TCP • Transfer Control Protocol • 3 step connection setup • Sends 1 packet, waits for acknowledgement • Sends 2 packets, waits for acknowledgement • Number of packets increases exponentially

35. TCP (cont.) • Number of packets increases exponentially until a threshold is reached • At this threshold, the number of packets sent reverts back to one and then increase linearly until a new lower threshold is determined • This second threshold is the connection speed

36. TCPGood for data, bad for voice • Too much delay in the 3 step setup • The acknowledgement confirms that data is transferred successfully, but creates too much delay for voice transfer

37. UDP • User Datagram Protocol • No connection acknowledgement • No transfer acknowledgement • Send packets and forgets them • Since 5% packet loss is acceptable for a voice communication, UDP is ideal

38. Basics of Telephony • Transmission • Transfer of data from one end to another • Switching • Converting data from one medium to another • Analog to Digital • Signaling • Setting up and tearing down a call

39. Making a Call • Dial a number • Sends a message to your Private Branch eXchange • Sends a message to your Central Office • Generates Initial Address Message • Connects to CO of receiving end • Sometimes through Tandem CO • Sends message to PBX of receiving end • Rings the phone on the receiving end

40. Making a Calling

41. Signaling • SS7 (System Signaling 7) • Allows a telephone network to setup route and control calls • 3 major components of SS7 • SCP (Signal Control Point) • Helps determine how to route a call • SSP (Signal Switching Point) • Starts, teminates and combines calls • STP (Signal Transfer Point) • Routes network traffic between switches

42. Signaling

43. Addressing • E.164 Addressing • NANP North American Numbering Plan • Area code • CO number • Line Number

44. Conclusion • FREE LONG DISTANCE CALLING! • Existing networks are barely able to carry VOIP communications • More developments compress the voice communication more and more, as networks keep improving • Possibly the new way to make phone calls?