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Voice traffic can wreak havoc with a PC's CPU, and process serialization has an impact on overall quality. Often our 200 MHz Pentium Pro PCs were heavily burdened by audio processing. This problem can be solved by introducing hardware on the desktop that offloads codec functions and performs several processes in parallel. Listen closely, and you can hear the difference with hardware assist and without it, using the same compression algorithm.

Read about and download sample WAV files:

Download and play packet loss samples:

• 10 percent packet loss/G.711 G71110.wav (602 KB)
• 20 percent packet loss/G.711 G71120.wav (512 KB)
• 50 percent packet loss/G.711 G71150.wav (497 KB)
• 10 percent packet loss/G.723.1 G72310.wav (632 KB)
• 20 percent packet loss/G.723.1 G72320.wav (592 KB)
• 50 percent packet loss/G.723.1 G72350.wav (776 KB)

Read technical notes

• Audio fidelity drops quickly under degraded network conditions, but the G.711 codec provides slightly better quality when the network loses packets. Listen for yourself.
• H.323 is a boon to the wiretapper. Listening in on a call is just a matter of using a LAN analyzer. Listen, and be warned.

Security is just an option under H.323. Suddenly, your LAN offers an easy way to listen in on conversations. This sample was recorded in real-time with RADCOM's AudioPro application running on the RADCOM PrismLite network analyzer.

Pretty scary, isn't it?

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This is a sample of NetMeeting's G.723.1 6.4 Kbps codec. Audio quality is good, although not as good as Quicknet Technologies' G.723.1 implementation . It also uses much more of CPU resources, typically 60% during our tests, 5 times the load of the Quicknet implementation.

At 6.4 Kbps, G.723.1 makes good use of bandwidth, but quality degrades rapidly with packet loss . When the network has appreciable loss, G.711 is a better choice.

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This is a sample of NetMeeting's 16Kbps codec from Lernout and Hauspie. In general, L&H's codecs provide low bandwidth utilization with moderate CPU load; 40% on our Pentium Pro 200. While the 16K version provides the highest quality audio of all the L&H codecs, we can't recommend any of them. A better choice is G.723.1

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This is a sample of NetMeeting's 8Kbps codec from Lernout and Hauspie. In general, L&H's codecs provide low bandwidth utilization with moderate CPU load; 45% on our Pentium Pro 200. Audio quality is poor, with noticeable distortion.

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This is a sample of NetMeeting's 12Kbps codec from Lernout and Hauspie. In general, L&H's codecs provide low bandwidth utilization with moderate CPU load, 38% on our Pentium Pro 200. Audio quality is poor, with a quavering, watery tone.

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This is as good as NetMeeting 2,1 is going to get. While this 64Kbps codec wastes bandwidth (remember that a call needs 64K in each direction!), this codec delivers good quality and slow degradation with packet loss. CPU utilization on our Pentium Pro 200 ran at 27%.

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Microsoft's ADPCM codec balances CPU utilization, network load, and quality. Too bad that it is mediocre on all counts. Audio quality is fair, with noticeable background noise. CPU utilization is moderate (24% on our Pentium Pro 200), and it doesn't save much bandwidth compared to G.711.

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NetMeeting's default settings make the CELP algorithm the codec of last resort, and we understand why. Audio quality is poor, CPU load is high (60% on a Pentium Pro 200), and it doesn't save much bandwidth compared to G.723.1

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Quicknet Technologies Internet LineJACK provides excellent quality when used with Quicknet's own H.323 application. It does this by offloading audio processing from the PC's CPU and performing digitization, echo cancellation, compression, and packetization in parallel. Our Pentium Pro 200 was barely touched by a G.723.1 call - with utilization hovering around 12%, as opposed to the 60% utilization we found with Microsoft NetMeeting.

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Quicknet's G.711 codec delivers excellent quality, indistinguishable to our ears from a regular PSTN call. Voices sound natural and line noise is low. CPU loading is moderate (27% on a Pentium Pro 200), and quality doesn't drop off as quickly under degraded network conditions as the G.723.1 codec.

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It's one thing to demonstrate voice over IP service when the network is clean, and quite another matter in the real world - where delay variation and packet loss can kill audio quality. While no codec can deliver toll quality when packet loss is high, the G.711 codec doesn't degrade as quickly when your network deteriorates.

In general, G.711 showed little degradation with 5% packet loss, but G.723.1 calls suffered. Compare what happens when the network loses 10% of its packets using the G.711 codec vs the G.723.1 codec . Neither sounds clean, but G.711 is slightly better. Note the twangy, metallic edge on the G.723.1 call.

At 20% packet loss, quality is poor with both codecs, although G.711 still beats G.723.1 . Note the substantial popping and dropouts in both samples. Voices on the G.723.1 call sound robotic.

At 50% packet loss, calls may be completed but the quality is unacceptable. This call should never be allowed on the network, but without a gatekeeper it is still allowed to ring through. Listen to G.711 and G.723.1. Remarkably, much of the audio is still understandable - a testament to H.323's resiliency. We could even disconnect the LAN cable and resume the call by reconnecting - which never happens with standard phones.

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