Presentation on Introduction to Telephony VoIP

Introduction to Telephony VoIP

1. Introduction to Telephony & VoIP Lesson 1 2013

2. Lesson Objectives After completing this lesson, you will be able to: Describe how digital signaling differs from analog signaling Explain the basic concept of voice over IP communications Describe the purpose of the gateway in a VoIP network

3. Digital Communication

4. Digital Communication A digital trunk is a single communication path between two switches that is used to carry many simultaneous voice conversations Remote Central Office Local Central Office

5. Pulse Code Modulation (PCM) A method of encoding an audio signal in digital format A standard audio signal is encoded as 8000 analog samples per second, of 8 bits each, giving a 64 Kbit/s digital signal known as DS0 The default signal compression encoding on a DS0 is either μ-law (North America and Japan) or A-law (Europe and most of the rest of the world)

6. Time Division Multiplexing (TDM) 64 Kbps 64 Kbps 64 Kbps 64 Kbps 1 2 3 . . . 32 1 2 3 . . . 32 1 2 3 Uses time-division multiplexing

7. E1 1 2 15 17 30 1 2 15 17 31 2.048 Mb/s Framing and Maintenance Signaling . . . . . . Voice Voice Voice Voice Voice 16 Data rate of 2.048 Mbit/s (full duplex) Split into 32 time slots Each time slot sends and receives an 8-bit sample 8000 times per second ( 8 x 8000 x 32 = 2,048 Mbit/s) Ideal for voice telephone calls where the voice is sampled into an 8 bit number (PCM) One timeslot (TS0) is reserved for framing purposes One timeslot (TS16) is often reserved for signaling purposes

8. T1 Frames 1.536Mb/s 1 2 F R A M I N G Voice Voice Signaling Data rate of 1.544 Mbit/s Split into 24 time slots each encoded in 64 Kbit/s streams 8 Kbit/s of framing information for synchronization 64,000 x 24 + 8 = 1544 Mbit/s Timeslot (TS24) is often reserved for signaling purposes

9. Signaling Methods Voice + Signaling Link In-band signaling is the exchange of signaling (call control) information on the same B-channel that the telephone call itself is using CAS (Channel Associated Signaling ) Out-of-band signaling is the exchange of signaling that is done on a channel that is dedicated for this purpose and separate from the channels used for the telephone call Common Channel Signaling (CCS) such as ISDN and SS7 Signaling Link Voice Link

10. ISDN Integrated Services Digital Network is an ITU-T term for integrated transmission of voice, video and data on the digital public telecommunications network Two interfaces are available: PRI (Primary Rate Interface) primarily used to link PBXs and to connect a PBX to the PSTN. Composed of 23 or 30 B-channels and one D-channel, all at 64 Kbps BRI (Basic Rate Interface) an ISDN interface typically used by smaller sites and customers. Consists of a single 16 Kbps D-channel plus 2 B-channels for voice and/or data

11. ISDN (Q.931) Call Flow Off hook, Dial Tone, Dialing Setup Call Proceeding Release Disconnect Alerting Connect Release complete Off hook Ringing On hook Calling Party Called Party Ringback Tone ISDN Digital Trunk Voice Channel

12. Clock Synchronization PBX Master Clock Toll Center PBX Timing Timing Timing End Office End Office Timing Timing

13. DTMF – Dual Tone Multi-Frequency 1 2 3 A 4 5 6 B 7 8 9 C * # D 1209 1336 1477 1633 697 770 852 941 DTMF is the common method of sending dialing information (replaced pulse dialing of the original telephone networks) Each number is represented by two tones which are transmitted simultaneously over the voice path Each row representing a low frequency and each column representing a high frequency

14. Call Progress Tone Description Dial Tone Indicates that the telephone exchange is working, has recognized an off-hook, and is ready to accept digits Ringback Tone This tone assures the calling party that a ringing signal is being sent on the called party's line Busy Tone Indicates to the calling party that the remote phone is occupied Reorder Tone (Fast Busy) Indicate that a person has dialed an invalid code, or that all trunks are busy and/or their call is unroutable Call Progress Tones In Telephony, call progress tones are audible tones sent from the PSTN or a PBX to calling/called parties to indicate the status of phone calls

15. Voice over IP (VoIP)

16. What is VoIP ? Voice over Internet Protocol (VoIP) is a set of technologies that enable the transmission of voice traffic over IP-based networks instead of the Public Switched Telephony Network (PSTN)

17. Circuit vs. Packet Switching Circuit Switching Traditional voice calls, running over the PSTN, are made using circuit switching, where a dedicated circuit or channel is set up between two points before the users talk to one another Packet Switching Data transmission technique in which data is separated into small 'packets', each with its own routing information and then sent through a shared, often public, network; at the other end the packets are reassembled into the original data format In this method bandwidth is only used when something is actually being transmitted

18. VoIP Protocol Stack VoIP is composed of two key components: Bearer (actual voice being sent over the network) using RTP/RTCP protocols Signaling (additional messaging that is necessary to control, establish, and tear-down the voice calls) The most common signaling protocols are: SIP H.323 MGCP MEGACO

19. RTP RTP (Real-Time Transport Protocol) is used to encapsulate VoIP data packets inside UDP packets RTP provides end-to-end network transport functions suitable for applications transmitting real-time data Sequence Number Time Stamp Synchronization Source ID - SSRC Voice Bits V P X M PT RTP Header 12 octets CC

20. Voice Codecs Codec Bit Rate (kbps) G.711 PCM (A-Law / Mu-Law) 64 G.726 ADPCM 16, 24, 32 and 40 G.729 CS-ACLEP 8 G.723.1 CELP 6.3 and 5.3 A codec (Coder/Decoder) converts analog signals to a digital bitstream, and back into an analog signal for transmission across IP networks Codecs generally provide a compression capability to save network bandwidth Some codecs also support silence suppression, where silence is not encoded or transmitted

21. VoIP Challenges Delay – Each component in the voice path adds delay (sender, network, receiver ). ITU-T G.114 recommends 150 msec as maximum desired delay to achieve high voice quality Jitter – Variation in delay; the effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing audio Packet loss – Occurs either in bursts or due to congested network. Periodic loss in excess of 5-10% of all VoIP packets can degrade voice quality significantly

22. Delay Packet X Transmitted Sender Receiver t Network Transit Delay Processing Delay Processing Delay End-to-End Delay Start Talk Packet X Arrive Start Hear Network

23. Jitter Jitter (delay variation) caused when voice packets suffer different transit delays, causing variation in arrival times at the receiver end The jitter buffer collects voice packets, stores them and sends them to the voice processor in evenly spaced intervals t t Sender Receives A B C A B C D 1 D 2 = D 1 D 3 = D 2

24. VoIP Gateways

25. Enterprise PSTN & Data Network Branch Headquarters Telecommuter PSTN IP Backup

26. IP Mediant 2000 PBX Mediant 1000 PSTN E1 / T1 IP Signaling IP Voice E1 / T1 PCM Digital Gateway

27. Media Processing