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 clientīs (Mic) input with a permanent tone i.e. a 440 Hz Sinus wave itīs 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 senderīs side does not fit the receivers rate. I do not understand this because both phones are using G.711 ALAW. Taking a look at *īs channels with the help of itīs 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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