Presentation on theme: "VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003."— Presentation transcript:
VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003
Page 2 Agenda Internet Telephony Call Basics Fundamental Components of VoIP Gateways VoIP Applications
Page 3 Voice Over Internet Protocol (IP) There are three styles of Voice over IP calls: Phone to Phone PC to Phone PC to PC Internet Intranet Gateways adapt traditional telephony to the Internet.
Page 4 Telephony Signaling Central Office Switch Idle First Digit is Dialed DTMF Detector Activated in the CO Dial-Tone On Remaining Digits Dialed Dial Tone Off Ring Back Voice-mode Connected On-Hook Off-hook Signals are exchanged between a telephone and the switch at the Central Office. These signals connect and disconnect calls as well as inform the caller of the progress of the call. Signals are exchanged between a telephone and the switch at the Central Office. These signals connect and disconnect calls as well as inform the caller of the progress of the call.
Page 5 Internet Intranet Packet Signaling All three VoIP calls can use H.323, or SGCP/MGCP to set up the Internet portion of the call. Calls involving gateways must also perform telephony signaling.
Page 6 Voice over Internet Signaling Sending voice over a data network requires advanced signaling techniques in the gateways. Internet Intranet Central Office Switch The gateway connected to the central office must emulate the telephone. The gateway connected to the phone must emulate the signaling functions of the central office.
Page 7 Voice over Internet Signaling Telephone numbers are translated to data network addresses (Internet addresses). Internet Intranet Central Office Switch Telephony signals are interpreted by the gateway and mapped to the appropriate network protocol (H.323/SGCP/MGCP for IP) set-up, maintenance, billing and tear-down messages.
Page 8 PBX Telephone DSPMICRO DSP Off-hook Voice-mode voice mode On-hook Idle-mode Off-hook DTMF Mode first digits Dial-tone digits idle mode Dial-tone off Dial-tone Switched CAS (FXS-FXO) Network connect connect_ack release setup call_proceeding connect connect_ack Call Progress In Band release H.323 SGCP/MGCP
Page 9 Fundamental Components of VoIP Gateways
Page 10 Micro Ethernet (Internet) Micro Processor(s) Telephony Protocols Network Protocols Management Routing Billing How is it all Done? Within the Gateway a series of processors perform the adaptation from Traditional to Internet Telephony. DSP Telephones (Circuits) Digital Signal Processor(s) (DSP) Voice Compression Tone Detection/Generation Echo Cancellation Silence Suppression
Page 11 Analog Voice to PCM An analog voice signal is received. The Signal is converted to a Pulse Code Modulation (PCM) digital stream DSP
Page PCM Processing The PCM stream is analyzed. DSP F Detected signaling tones are routed around the CODEC. (needed, since most CODECs garble signaling tones to the point that they are unrecognizable) Tone Detection is performed: Echo is removed. The Voice Activity Detector (VAD) removes silence. Remaining stream is passed to CODEC.
Page 13 PCM to Frames and voice frames are created Most CODECs also compress the PCM stream: PCM G.711 generates 64,000 bits per second G.729a compression generates 8,000 bits per second DSP The PCM stream is fed into the CODEC Each Frame is 10 ms long (G.729a) and contains 10 bytes of speech.
Page 14 Frames to Packets DSP Packet Assembler Software within the DSP takes frames from the CODEC and creates packets. The packet is forwarded to the gateways host processor. Several frames may be combined in a single packet RTP A 12 byte Real Time Protocol (RTP) Header is added: Provides sequence number Time stamp
Page 15 IP A 20 byte IP header is added to the packet containing: The IP address of this gateway (the source address) The IP address of the destination gateway An 8 byte UDP header containing source and destination sockets is also added. UDP Addressing Dialed digits identified by the tone detection performed in the DSP are used to determine the destination number RTP This number is mapped to an IP Address. = Micro
Page 16 In the Internet Routers and Switches in the Internet examine the addresses in the IP address in order to identify the route to the destination. Several routers and or switches may be in the path that the packets take to their destination.
Page 17 IP Upon Arrival at the Destination The IP and UDP headers are removed from the packet in the Microprocessor. UDP Micro RTP The Packet is forwarded to the DSP where the RTP Header is removed. Finally, the packet is disassembled leaving the voice frames
Page 18 Various Network Problems are Dealt With Voice Packets are generated at a constant rate while someone is speaking; there is essentially no gap between packets. These gaps, known as jitter, must be removed by the receiving gateway in order to accurately reproduce the original speech Devices in the network cause an unpredictable amount of delay to occur between packets.
Page 19 Jitter Removal An adaptive jitter buffer in the receiving DSP is used to smooth the playout of packets arriving from a jittery network. DSP This eliminates the jitter induced distortion that would have been heard by the listener.
Page 20 Lost Packets Congestion in the network may cause some packets to be dropped Left untreated, the listener hears annoying pops & clicks.
Page 21 Lost Packets An algorithm in the DSP detects missing packets And replays the last successfully received packet at a decreased volume in order to fill the gaps DSP 35
Page 22 Turning Hello….. Into oHell Out of Order Packets Out of order packets are not played in the order they arrive….. Packets may take diverse routes through a network and may arrive out of order DSP
Page 23 Out of Order Packets DSP When an out of order condition is detected the missing packet is replaced by its predecessor as if it is lost. 2 When the late packet finally arrives it is discarded. 3
Page 24 PCM Back to Analog A Comfort Noise Generator fills in the gaps that were created during silence detection and suppression. The PCM Stream is reconstituted as an analog signal and is played out to the listener
Page 25 VoIP Applications
Page 26 Central Office/Infrastructure Central Office Gateway Packet Network Traditional carriers migrate to packet core for lower network costs. Gradual capping of Class 4 tandem switches drives CO/Infrastructure VoIP ports. Carriers proposing new packet architectures with dramatically lower cost structures.
Page 27 Enterprise SME Gateway IP Phone PBX Packet Network Enterprises deploying to avoid access charges and settlement fees. Businesses take advantage of existing data networks. Reduced operating costs by managing one network.
Page 28 IP Phones and PBX Trunking Office 2 Office 1 T1 Packet Network Gateway Router IP Phone PBX LAN-based PBX for cost reduction, flexibility, and new applications: Integrated voice/data LAN infrastructure Integrated voice/data applications Open hardware platform
Page 29 Residential Broadband Residential voice alternatives, leveraging broadband connections VoCable solutions in trials in US, and deployments in Europe VoDSL deployments in Asia and Europe Fiber to the Home potential in China CMTS DSLAM CPE Gateway Cable Modem Packet Network Cable or DSL Modem Based IAD Voice Gateway
Page 30 Summary VoIP solutions require well integrated, robust set of functional components for toll quality operation. VoIP implementations are in current systems deployed worldwide. VoIP value proposition exists in different vertical markets.