Designing Apps using DSP s. Sandeep Harpalani. Residential Gateway market. Analog Devices. - VoIP Applications for

Transcription

1 Designing Apps using DSP s - VoIP Applications for Residential Gateway market Sandeep Harpalani Analog Devices

2 The Changing Market Landscape Voice Circuit-Switched Data Packet Routing Entertainment Broadcast

3 IP Convergence Converged Network to Deliver Triple Play Video, Voice and Data to Connected Clients Multi-service Solution Wired/Wireless Access Voice Circuit-Switched to Voice over Network Wire-speed Performance Data Packet Routing Switching and Routing Quality of Service Broadband Transport Entertainment Broadcast to Digital Media Security and Firewal l

4 Next Generation Converged Residential Solutions Entertainment Video Streaming Video Conferencing Gaming Music Personal Computing Data Routing Broadband Access Sharing Residential Gateway Communications: VoIP Fax GSM Integration PDA Integration Home Automation Security RFID Integration Remote Appliance control Gateway is the key element for delivery of new broadband services and revenue streams to enable the Service Provider to exploit the Broadband Access

5 Human Interface Speech Recognition Text To Speech Handwriting Audio Wired Connectivity USB TCP/IP Ethernet Operating Systems / RTOS What Does DSP Enable? Micro- Processing Digital Signal Processing Image Processing System Control / Applications Software Wireless Connectivity Bluetooth GSM / GPRS 3G / EDGE Digital Imaging CODECs MPEG JPEG H.263 H.264 Designed for High Level Language

6 VoIP Applications with DSP IP-PBX s Desktop IP Phones Wireless LAN IP Phones VoIP Router Analog Telephone Adapter (ATA) Residential Gateway Integrated Access Devices IP-Set Top Box Video-Conferencing Voice Processing Boards Audio-conferencing Satellite, Microwave, Mobile and Rural Radio Network

7 Application Trends Converging Multimedia and Network Communications Market demand for multiple formats Video: WMV ver.9, H.264, MPEG-4, MPEG-2, MJPEG Audio: WMA PRO ver.9, MP3, MP3 PRO, AAC, Dolby Digital, DTS Speech: G.711, G.728, G.729AB, G.723.1A, AMR Wireless Communications: GSM/GPRS, EDGE & 3G Fixed ASICs can not offer the cost, flexibility or time-to-market for multi-formats Conventional programmable architectures can not meet demanding signal processing requirements

8 Packet Processing Pipes are getting larger Access Gateway LAN ADSL2+ VDSL VDSL2 WiMax EPON GPON 802.3,802.11g BT HPNA HomePlug ETC

9 Packet Processing Rich Media Support with Real Time requirements Access Gateway LAN ADSL2+ VDSL VDSL2 WiMax EPON GPON 802.3,802.11g Bluetooth Firewire HomePlug routing bottleneck lost packets

10 LBR Channels Lifeline Residential Gateway IP-Set Top Box Ethernet LAN Ports USB 2.0 Device Bluetooth SPI 10/100 Eth Switch MII SLICs & Codecs PCM/ SPORT Network Processor & DSP Utopia ADSL2/2+ Phy Powerline PHY MII PCI WLAN a/b/g USB 2.0 Host Powerline WAN Uplink

11 Choosing The Right Gateway Processor Architectural changes required o Speed want to achieve the performance commonly associated with full-hardware switching and routing implementations Real-time Requirements o Scalability Enough headroom to handle new requirements New Applications: Home Automation, Entertainment Security Applications: Content Filtering, Access Control o Flexible want the flexibility to meet challenges not afforded by dedicated hardware solutions IPv6 New security protocols Multiple Peripheral Interfaces

12 Choosing the Right DSP in VOIP? Features include o Low Power o High Speed o Small Footprint o Rich Instruction set o Single Cycle Execution o External memory access o Scaleable DSPs (also Pin Compatibility) o Interfacing multiple DSP Processors o Integrated DSP Processors with Network Processors o Real Time Voice Processing solution: Distributed Processing

13 VoIP Architecture

14 TIP RING Call Origination Point S L I C On/Off Hook States Active Standby Ring VoIP Design Flow CODEC Host Processor SPI Signaling DSP Voice Processing SPORT/SPI Commands WAN Interface ADC Voice PCM SPORT0 Voice Packets DAC Ring Generation GPIO s GPIO_ISR GateKeeper/ Media Gateway Controller/ Proxy Server MII Utopia x PCI/PCMCIA LAN LAN Call Destination End Point

15 DSP SOFTWARE Encoder 711/729AB/ 723.1A DTMF VAD Fax Demod IP Voice Architecture PROCESSOR SOFTWARE TO HOST Host Interface PROCESSOR To DSP Processor SPI Interface of SLICs, CODECs POTS-Call States information through SLICs and CODECs Fax Mod Decoder 711/729AB/ 723.1A CNG PLC Tones and Caller ID AJB MGCP SIP RAS H323 Control H225 H.245 G.168 Echo Canceller RTP/RTCP SIP on UDP or TCP PCM Interface UDP TCP IP CODECs and SLICs To Physical Layer

16 IP Voice Software Modules DSP Software: o VoIP Software Modules PCM Interface Tone Detection and Generation (DTMF/ Call Progress Tones) Echo Cancellation Voice Compression: Support for Voice Codecs (G.711/G.723.1/G.729AB) Voice Activity Detection/Idle Noise Measurement for Comfort Noise Generation (CNG) Jitter Removal Voice Packetization/ Depacketization Fax-Passthrough/Fax Over IP (T.38)

17 DTMF VoIP Architecture: DSP To Host Processor Host Interface Speech Encoder Fax Demod F ax Speech Mo d Decoder Comfort Noise Generation Packet Loss Concealment Voice Activity Detection (G711/G726) G.168 Echo Canceller PCM Interface CODECs & SLICs Tones & Caller ID

18 Speech Codec Voice Codecs Compression Technique Bit Rate Frame size G.711 PCM 64.0 Kbps msecs 4.4 MOS 1 G.729 Conjugate structured Algebraic Code 8.0 Kbps 10.0 msecs 4.13 Excited Linear Prediction (CS- ACELP), G.729B for VAD option. G723.1 Multi-Pulse Maximum Likelihood Quantization (MP-MLQ) 6.3 Kbps 30.0 msecs 3.94 G.726 Adaptive Differential Pulse code Modulation (ADPCM) 32.0 Kbps msecs 4.0 G.729A Conjugate structured Algebraic Code Excited Linear Prediction (CS- ACELP), G.729AB is with VAD option 8.0 Kbps 10.0 msecs 4.09 G Conjugate structured Algebraic Code Excited Linear Prediction (CS- ACELP), G.723.1A is with VAD option. 5.3 Kbps 30.0 msecs 3.77 G.728 Low Delay Code Excited Linear Prediction (LD-CELP) 16.0 Kbps msecs 4.22

19 VoIP Architecture- Host Processor Voice Algorithms DSP Processor To SLICs, CODECs POTS-Call States information through SLICs and CODECs DSP IPC RTP/ RTCP MGCP SIP Adapti ve Jitter Buffer SIP on UDP or TCP H.323 UDP TCP IP (With QoS) To Physical Layer

20 Voice Quality Enhancements

21 Issues with Real-time Voice in IP Networks Delay Jitter Echo: Round trip Delay should be < 50 millisecond. Requires Echo Cancellation End to End Delay: Talker Overlap/Double Talk. End to End Delay should be < 250 millisecond Buffer to handle Jitter Packet Loss Interpolate for Lost speech packets Duplicate Packets

22 Causes For End to End delay CPE Specific Delay: o Codec selection o Packetization interval o System Architecture hardware & software OS dependencies o Low Latency by IPQos Implementation Network Specific Delay o Propagation in the physical IP media o Network impediments can increase end to end delay o End to end available bandwidth

23 Robust Echo Canceller Algorithm Attributes Should Comply ITU G168 Recommendations at minimum o Quick Convergence and re-convergence for different Echo path models. o Good Echo Removal (ERLE) performance o Robust Double Talk detection o Good System Hooks for Voice Quality Monitoring, log different parameters, and to apply feedback Requirements Beyond G.168 o Field conditions Background conditions, close-by speakers creating acoustic coupling o Varying phone types, parallel 2-3 phones on the same RJ-11 Jack

24 Clock Drift Issues Clock drifts of end boxes can create excess or less packet at regular interval Jitter buffer takes care of excess or less packets PLC also helps in case of less packets or packet drops in the network Packet Loss Concealment (PLC) Algorithms for all voice coders. Clock PPMs, and PPM corrections in association with Jitter buffers will improve voice quality

25 Voice Playout Algorithms The jitter buffers add delay, which is used to remove the packet delay variation that each packet is subjected to, as it transits the packet network. The voice playout algorithm basically contains the jitter buffer management. In adaptive Jitter Buffer scheme, buffer size is dynamically varied, depending on the past statistics of arrival times. The Adaptive Jitter Buffer size changes dynamically with the varying network conditions. This also arranges the out-of-sequence packets in the sequence. Duplicate packets are discarded. It has to take care of silence vs packet loss. It has to take care of Bursty spikes also.

26 Fax over IP Low bit rate codecs G.729A, G don t support Fax, Switching to G.711 is one of the option on sensing Fax call FAX: Switches to either G.711 or T.38 T.38 reduces bandwidth requirements, works better for Fax with lot more redundancy Usually VAD/CNG, EC, PLC, Adaptive operation of Jitter buffer are disabled in Fax operation T.38 takes care of packets and raw payload and modulation/demodulations are taken care by V.21, V.27, V.29, V.17 Supports Fax from 2400 to bps FAX quality and speed of operation also depends on Front end SLIC-CODEC.

27 Telephone Front End SLICs Each RJ-11 port to be capable of driving 2-3 phones (REN) Matching Country impedance Reasonably tolerate multiple country phones o Independently configure each port Should support Call feature extended functions Lifeline capability SLIC-CODEC should allow Wideband capability Tr-57 transmission characteristics Able to configure Gains, Country specific loudness ratings through Gain/loss planning GR-909 to sense remotely the conditions of SLIC remotely Engineered w.r.t self Health monitoring and relaying the statistics on network based Logger etc. Protection for wrong usage Instead of telephone, Wall RJ 11 may get connected-both giving higher voltages

28 VoIP-Voice Required Attributes End to End delay reduction IPQOS implementation for providing voice priority Robust Echo cancellation Algorithm design. Adaptive Jitter Buffer management Algorithm. Packet Loss concealment Algorithms for all voice coders. Clock drift PPMs correction in association with Jitter buffers Better telephone front end, impedance matching, drive for parallel phones Fax machines operation on VoIP through Pass-through and T.38 DTMF inband out band and delays Capability of upgrades to Wideband Codecs Capability for call features Well Engineered w.r.t self Health monitoring and relaying the statistics on network based Logger etc.

29 The End