Maybe a NAT issue?



On Thu, Sep 12, 2013 at 5:07 AM, Vincent Xia <gmangudai@gmail.com> wrote:
im trying to encrypt the voice call over FS in terms of SIP and RTP stream, what i do is simply follow instructions in http://wiki.freeswitch.org/wiki/SIP_TLS, the soft phone im using is PhonerLite 2.10, now the call establishment is OK, but after the called 
party answers, there's no voice at both sides, no RTP packets can be seen in wireshark which runs at FS box. 

The PhonerLite works fine with FS in uncrypted mode, so what's the problem with my configuration of SIP TLS?


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