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LatencyMeasurementsMain.LatencyMeasurements HistoryHide minor edits - Show changes to output Changed lines 22-23 from:
This was played on the same laptop used for recording. The original reference source can be to:
This was played on the same laptop used for recording. The original reference source can be [[http:/wiki/misc/latency | found here]] Changed lines 43-51 from:
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1. PSTN to PSTN: no appreciable echo could be found. 2. PSTN to cell: 135ms delay between signals 3. Cell to cell: 270ms delay between signals 4. OpenPhone to SIPP: 260ms delay between signals 5. Internal audio loopback (no SIP protocol) : 180ms delay between signals !!Analysis The reference tests (1 through 3) are consistent with real life experience. Test #5 shows that latency within the audio device being used is approx 180ms. This is atrocious, but PC sound cards, and Vista, are designed for high quality audio streaming, not interactive voice applications. If the audio latency is subtracted from the results of test #4, it appears that the latency within Opal is around 80ms. This is expected given the requirements of the jitter buffer - any less and audio could break up if the network traffic started jittering. Changed lines 13-14 from:
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* Audio loopback device (no network or protocol stack used) Changed lines 13-14 from:
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* Audio loopback device Added line 51:
* Audio loopback device: 180ms delay between signals Changed lines 12-13 from:
* OpenPhone on Windows to SIPP in loopback mode to:
* OpenPhone on Windows to SIPP in loopback mode (G.711 codec) Changed lines 38-39 from:
The recorded WAV files can be [[/wiki/misc/latency | found here]] to:
The recorded WAV files can be [[http:/wiki/misc/latency | found here]] Changed lines 23-24 from:
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!!Methodology Changed lines 36-49 from:
The recorded For each call, the time was measured (using CoolEdit) from the start of the direct audio and any echo or !!!PSTN to PSTN to:
!!Results The recorded WAV files can be [[/wiki/misc/latency | found here]] The results were as follows: * PSTN to PSTN: no appreciable echo could be found. * PSTN to cell: 135ms delay between signals * Cell to cell: 270ms delay between signals * OpenPhone to SIPP: 260ms delay between signals Changed lines 27-28 from:
remove unwanted acoustic sources or to reduce echo. to:
remove unwanted acoustic sources or to reduce echo. This test showed there was no appreciable echo past 40ms after the start of the original tone. This was considered acceptable for testing. The measurement methodology was to assume that any "echo" present after 50ms of the start of the tone was a recording of the received signal. The time between the start of the direct tone and the received tone was then measured using CoolEdit. The recorded results can be found here: For each call, the time was measured (using CoolEdit) from the start of the direct audio and any echo or Changed lines 12-13 from:
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* OpenPhone on Windows to SIPP in loopback mode Changed lines 20-22 from:
The audio source was a the originating endpoint to:
The audio source was a WAV file containing five 1ms audio pulses, spaced at 0.75 seconds, at -3db. This was played on the same laptop used for recording. The original reference source can be found here Added lines 25-28:
Before starting the tests, a reference recording was made of the environment, which was a small office containing a large number of computer systems. No steps were taken to remove unwanted acoustic sources or to reduce echo. Added lines 1-27:
(:notitle:) !Measuring Voice Latency !! Introduction This document describes some measurements of the latency of various types of voice calls. The call types measured were: * PSTN to PSTN * PSTN to cell phone * Cell phone to PSTN * Cell phone to cell phone In each case, the latency was measured using CoolEdit 96 on a laptop to simultaneously record the audio fed into the microphone of the originating endpoint and the audio departing the speaker of the destination endpoint. All audio was recorded at 48khz mono using 16 bits per sample. The audio source was a small pieces of metal tapped sharply together near the microphone of the originating endpoint. !!Results !!!PSTN to PSTN |