My test server with Flashphoner Web Call Server is connectd to a SIP trunk.
Connection establishes fine but it lack audio completely. Codeс is G.729. RTP streams run fine in both sides, back and forth. I looked at logs but can't figure out anything from them. What could be the problem?
Connection establishes fine but it lack audio completely. Codeс is G.729. RTP streams run fine in both sides, back and forth. I looked at logs but can't figure out anything from them. What could be the problem?
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