[Asterisk-Users] Frequency Shift

[Siegel, Joerg]

Hello,
I am using * as a SIP proxy with several SIP clients. The SIP clients are
SJPhone Soft phones. All clients are inside a firewall and the Server is
inside too. All is working fine, but the speech sounds like Micky Mouse. If
you feed one client´s (Mic) input with a permanent tone i.e. a 440 Hz Sinus
wave it´s frequency on the (Speaker) output of the client you are connected
to is shifted to a higher frequency. In addition to this you can hear drop
outs. Obviously the samling rate on the sender´s side does not fit the
receivers rate. I do not understand this because both phones are using G.711
ALAW. Taking a look at *´s channels with the help of it´s command line
interface shows ALAW for both channels too. I set reinvite=no in the
sip.conf file, because SJPhone did not support this and the connection broke
down. So if I understand things right the conversion error could also be
caused by *, because it stays inside the rtp connection.

Does anybody know something about this phenomena??

Thanks in advance

joerg.      

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