ip phone design considerations
From: Apparatus (apparatus.home_at_lycos.com)
Date: 06/27/04
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Date: 27 Jun 2004 02:10:15 -0700
Hello,
I am planning on designing an IP phone for a student project. I would
like some advice on details that I need to take into consideration.
My basic thought is this: Use an ADC to take 12-bit samples of the
microphone at 8kHz. Next encode the samples into u-law g.711 PCM. Next
transmit over UDP packet. Next decode from u-law. Next send to 12-bit
DAC connected to PGA or op amp, then to speaker. Note that I am
planning on using a DSP on both ends, probably a Microchip d*** due
to the affordability of the development tools.
The concerns I have are mainly on the analog ends.
1) How much of a problem will noise be with this scheme? Should I add
a BPF before encoding the remove frequencies outside 400Hz to 4kHz?
How else should I remove noise?
2) Is interpolation on the receiving end needed to achieve a toll
quality signal? Can this interpolation be a simple capacitor or coil?
Will the speaker be a good enough interpolator?
3) In general, how is a microphone connected to a ADC? A DAC to a
speaker?
4) Is UDP a reliable enough transmit method? Should I add a 100ms
buffer for frame delays? Should I repeat the last frame if a frame is
omitted?
5) What else should I consider? Any other suggestions?
Cheers.
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