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Hi. I’m facing a really bad voice quality when a
make calls between a tdm user and a sip user. Take a look at the following scenario: sip-user ----> asterisk ----> TDM22B(fxo) ---->
PABX and PABX ----> my-tdm-extension When the sip-user places a call to my-tdm-extension, the
call goes through the TDM22B followed by the PABX and then I answer it in
my-tdm-extension. For the sip user the quality of the voice is normal,
but for the tdm-extension it’s unacceptable. I got some sequences of choppy/picotted
voice. The invert situation is also true, even if the
tdm-extension place the call to the sip user, the voice also is terrible for de
tdm side. First it looked like a problem with bandwidth but calls
between the sip-user and another sip user (this “another sip” is in
the same building that the tdm-extension is) are excellent, so this tells me
that bandwidths isn’t my problem. My PABX extensions group work pretty well among them
selves so look like that isn’t the problem either. I really think it is something with IRQ misses or
some bus problem but I’ve already followed the steps mentioned in
voip-info.org to test IRQ misses and I’m still unable to figure out what
is the problem. I’m using the GSM codec on the sip-user, but
even with ulaw the problem persists. Any help would be appreciated. Thanks, ---- Filipe Mordhorst |
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