I have noticed that most mobile phones (GSM and CDMA at least) seem to have a tendency to interrupt the incoming audio stream when the microphone levels get louder than a certain threshold (such as when you are speaking into it). I do not know exactly why this happens, nor whether it is something that happens in the handset (the handset mutes the incoming audio, which doesn't make much sense) or in the network (maybe to try to save bandwidth?).
Also, the GSM codec is intended to intelligibly encode human speech, not general audio signals such as music. GSM uses approximately 6-8 times less bandwidth than G.711u/a, and the degradation in the quality of music and other non-speech signals is one of the tradeoffs that we make in order to not have the cost of mobile calls increase 6-8 times over.
Maybe if Europe was as backward with its mobile systems as the United States currently is, you could try to get your users to call your system from analog mobile phones... They don't use digital compression (by definition, I suppose) and would probably work much better for your specific application ;).
-Rusty
On 11/25/05, Esteban Maestre <[EMAIL PROTECTED]> wrote:
Hi all!
I have a strange problem when using asterisk. I have configured asterisk
to receive calls (FX0). In my configuration, I want asterisk to play music
while I record the caller's speech. If the caller does the call from a
fixed line telephone, there is no problem, but in case the caller does the
call from a mobile GSM phone, the quality of the music he hears becomes so
bad, and even more when he speaks.
I have tried several codecs.
Any idea or advice?
thanks in advance,
-esteban-
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