Presentation on Introduction to Telephony VoIP
- 1. 1 Introduction to Telephony & VoIP Lesson 1 2013
- 2. 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
- 4. 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. 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. 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. 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 0 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. 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. 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. 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. 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. 12 Clock Synchronization PBX Master Clock Toll Center PBX Timing Timing Timing End Office End Office Timing Timing
- 13. 13 DTMF – Dual Tone Multi-Frequency 1 2 3 A 4 5 6 B 7 8 9 C * 0 # 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. 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. 15 Voice over IP (VoIP)
- 16. 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. 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. 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. 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. 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. 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. 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. 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 D1 D2 = D1 D3 = D2
- 24. 24 VoIP Gateways
- 25. 25 Enterprise PSTN & Data Network Branch Headquarters Telecommuter PSTN IP Backup
- 26. 26 IP Mediant 2000 PBX Mediant 1000 PSTN E1 / T1 IP Signaling IP Voice E1 / T1 PCM Digital Gateway
Editor's Notes
- #7: A digital interface running at 2.048 Mbit/s called E1 interface ITU-T recommendation G.732. Two types of physical interfaces may be used - the 75 ohms unbalanced coaxial interface or the 120 ohms balanced twisted pair interface (G.703). The actual data bits are transmitted using alternative Mark Inversion (AMI) with High Density Bipolar (HDB3) encoding.
- #8: T1 is common to North America and Japan, It offers 23 or 24 traffic timeslots depending upon the type of signaling being used. T1 uses Alternate Mark Inversion (AMI) line coding to eclectically encode the signal on the line and to avoid long strings of ‘0’ Also a process called B8ZS is also used to overcome the low density of AMI. Common Channel Signaling on DS-1 utilities. Timeslot 24 to carry signaling information as HDLC based data message.
- #9: There are two different signaling methods. The first method is In-Band, which means that the signaling is sent within the voice channel. The second method is Out-of-Band. This technique uses a separate channel for signaling. This is done, for example, in the ISDN protocol by D-channel, which is dedicated only for control while the voice is sent in B (bearer) channel alone.
- #13: Notes: Dual Tone Multi-Frequency will be covered in more detail in another lesson.
- #19: This Is a typical voice packet. Notice the total overhead (40 bytes)! RTP Header: The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. The timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations. The SSRC field identifies the synchronization source. This identifier is chosen randomly, with the intent that no two synchronization sources within the same RTP session will have the same SSRC identifier.
- #20: This table shows the ITU Standards including the bit rate in kilobits and the frame size per m/s. In evaluating the performance of codecs, several factors come into play: Frame size - also called frame delay, represents the length of the voice traffic measured in time. For example in PCM 1/8000*1000=0.125 fr/ms. Processing delay – also called algorithmic delay. This factor represents the delay incurred at the codec, to run the voice and coding algorithm on one frame. Look ahead Delay – occurs when the coder examines a portion of the next frame to provide guidance in coding the current frame. Frame length – the number of bytes resulting from the encoding process. DSP MIPS and required RAM are two hardware elements which specify the encoding process.
- #25: Traditionally, there are two isolated networks for telephony. One is called PSTN (Public Switched Telephone Network) also referred to as POTS (Plain Old Telephone Service), which is the total collection of interconnected voice-oriented public telephone networks. The second network is the IP Network, which refers to the Internet or the Intranet using a set of protocols called TCP/IP (Transmission Control Protocol/Internet Protocol).
- #26: Digital gateways convert (in real time) ISDN or CAS signaling to SIP and PCM to RTP.