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Netergy Networks’ Perspective Open Issues, Future Work. Mack hashemi Netergy networks Stanford university March 7, 2001. Motivation. Audio quality in a packet-switched network depends on: Latency Delay variations (jitter) Lost packets How to overcome lost packets?. Outline.
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Netergy Networks’ PerspectiveOpen Issues, Future Work Mack hashemi Netergy networks Stanford university March 7, 2001
Motivation • Audio quality in a packet-switched network depends on: • Latency • Delay variations (jitter) • Lost packets • How to overcome lost packets?
Outline • Overcoming lost packets • Netergy’s solution • Current implementation • Practical issues • Results • Open issues • Future work
Techniques to Overcome Lost Packets • Improve network infrastructure • Transmit more packets • Add redundancy • Alleviate losses
Improve Network Infrastructure • Research efforts have aimed to improve qos (e.G. RSVP) • Extensions to current protocols and switching algorithms to provide the desired requirements. • Problems: • New products not deployed in the internet. • Algorithms not implemented in the internet.
Transmit More Packets • Closed loop (e.g. ARQ) • Problems: • Adds to end-to-end delay • Not scalable (multicast) • Open loop (e.g. Path Diversity) • Problems: • Delay variation among paths eliminates benefit. • Requires infrastructure support, not in place.
Add Redundancy • Multiple Description Coding • Send multiple descriptions of the source over the unreliable link. • Hope that at least one of the descriptions can be received correctly. • FEC • Send a redundant packet every n packets. • Can recover from a single loss in n packets.
Alleviate Losses • Adaptive Play Out (more later…) • Sub-sample Interpolation • Odd and even samples are sent in different packets. • Information in the lost packet is recovered by interpolation from data in received packet. • Works well only with single packet losses.
Past Work: Adaptive Play Out • Netergy’s current implementation of Adaptive Play Out Algorithm: • Require a buffer for storing compressed data. • Require a buffer for storing raw audio data after decode. • Memory is premium on current T2-based systems, no raw audio data buffer. • Adaptive play out algorithm operates on the compressed data buffer.
Past Work (continued) • Some Results: • For the trace stanford_chicago.20long.one, 9000 packets at 20ms each
Past Work (continued) • Other practical issues with the media traffic • G.729 packetization could be a major source of confusion (different packet sizes, no in-band information about the type of frame) • Response time of some VAD algorithms is long. Packets may bunch up at the receive end after a silence period. The Adaptive Media Play out algorithm needs to handle this.
Past Work (continued) • Other issues regarding implementation in Netergy’s T2-based systems: • Time stamps are not reliable • The mechanism will work best if all the decoded data is stored in a buffer and the algorithm is allowed to work on extending/shrinking portions of the data. • The system loop operates on a 10ms timer. We need to decode only 10ms of audio and give it to driver for play out.
Past Work (continued) • The system has to deal with a wide variety of packetizations. The proposed algorithm has been tuned for packetization of 20ms. • Proposed algorithm modifies the time scale of the audio signal. This will result in poor scores given by standard speech quality measurement algorithms (e.g. PSQM), since they rely on synchronization between input and output.
Future Work • Use dynamic sizing of the compressed data buffer • Needs investigation and testing. • Analyze additional memory to accommodate increased buffer size: • Needs investigation and testing. • Needs cost analysis. • Study different packetization periods.
Future Work (continued) • Other: • Add redundancy in the packets in order to overcome lost packets. • Make use of the neighborhood statistics for the data to be decoded. • Incorporate adaptive techniques to further improve signal quality. • Wireless ???
General Open Issues • Design techniques to enable graceful degradation of signal on the presence of Packet losses. • Not clear how to choose the best redundant information. • Need to be able to adjust the rate at which packets are sent into the network and the amount of redundant information added in these packets.
Conclusions • Adaptive Media Play Out appears to be able to greatly enhance audio quality in presence of packet loss. • Need further analysis, investigation and testing to fully implement Adaptive Play Out Algorithm. • Continue to search for other ways to improve audio quality.