A Software Defined Radio Implementation for Voice Transmission over Wireless Ad-hoc Networks Jason Tran SURF-IT 2009 Fellow Mentors: Dr. Homayoun Yousefi’zadeh Dr. Hamid Jafarkhani Graduate Student: Ala Khalifeh

Introduction – Most radios today can only serve as one kind of radio. Cell phones  cell phones Walkie talkies  walkie talkies FM Transmitter  FM Receiver – Imagine a radio that can be programmed. – The Software Defined Radio (SDR) Example: Gobi - Qualcomm’s SDR can switch between EVDO (Verizon/Sprint) and HSDPA (AT&T)

My SURF-IT Research We’ve implemented a voice transmission scheme over a software defined radio platform using the Universal Software Radio Peripheral (USRP) programmed by GNU Radio. The goal of this project was to optimize voice transmission using voice codecs and forward error correction (FEC). These will be discussed in more detail later.

Motivation/Definitions Why Reprogram the Walkie Talkie? Fundamentally we are going from Analog -> Digital (like our new TV broadcasts). Three Advantages: – Efficient: less data is needed to be sent (compression) – Reliable: digital can overcome errors/noise – Flexible: no needed extra hardware to change signals which could save money

Analog vs. Digital What’s the Difference?! We only care about information at certain points. Information is now “discrete” or “digital”, but still transmitted in analog. Converting to Digital (aka Sampling)

Our Software Defined Radio (SDR) Our SDR = GNU Radio on Linux (Software) + Laptop USRP GNU Radio is an open source Linux software used to program the USRP. The USRP consists of digital to analog converters and a radio frequency front-end that sends the data over the air. +

Network Modes Intranet (Infrastructure) Ad-Hoc

The Project (Streaming Voice) We stream by packetizing a certain amount of data. – For example 6000Bytes 1500Bytes1500 Bytes or 6KBytes 1500 Bytes1500 Bytes Two usual transport protocols used for streaming are transmission control protocol (TCP) and user datagram protocol (UDP). – TCP: sends a packet and waits for acknowledgement before sending another – UDP: continuously sends without acknowledgement

Transmission Diagram

Transmission Errors Two types we are concerned about: – Packet Erasures (late or lost packets) – Bit errors Bit Errors (flipped or unknown bits) – Packets [3 byte example(8 bits per byte)] | | x010x1|x0010x00|10x010x0 Regular Packet Packet with Errors One Byte

Error Handling Voice Codec: Speex (data compression) – Uses less bandwidth and reduces packet erasures – Variable Bitrate and Echo Cancellation Forward Error Correction (FEC): Reed Solomon Codes – Creates redundant bits: Received “000” is seen as “0”“110” is seen as “1” Received “010” is seen as “0”“011” is seen as “1” With these, we’re able to optimize voice streaming.

Additional and Future Work This project is one part of a larger project of delivering multimedia content in real-time (voice, audio, video) and non-real time (HTTP and FTP) on a Mobile Ad-hoc Network (MANET).

Applications Mobile ad-hoc networks (MANETs) can establish a network where there is no infrastructure – Military (Joint Tactical Radio System (JTRS) ) – Emergencies – Medical SDRs enable“cognitive” radios – Self-adjusting radios

Thank You! Thank you SURF-IT and UROP coordinators! Questions?

Future Work and Applications Other current applications include cellular GSM base station, GPS receiver, FM transmitter/receiver, digital television decoder, and passive radar.

Metrics for Measuring Transmission Performance Packet Loss/Throughput Perceptual Evaluation of Speech Quality – Family of standards for a test methodology of evaluating speech quality. – Compare speech before and after transmission. – Mean Opinion Score (MOS) Packet Delay Variation – Jitter – IP Packet Delay Variation for IP Performance Metrics (IPPM)

Background Traffic (200/400/600/800kbps) Performance Analysis We know what link speed we will choose (1Mbps) In real life, we’ll have a lot of background traffic. We want to measure performance of voice stream under different network before and after using error handling schemes (encoder/FEC). Experiment setup: Measure voice transmission from computer 1 to computer 2. Voice Transmission Computer 3Computer 2Computer 1

Throughput vs. Raw Bitrate

Packet Error Rate (PER) vs. Raw Bitrate

Mean Opinion Score

End to End Delay

Jitter

Networking and the OSI Model Linux Reed Solomon Codes GNU Radio USRP