[Freeswitch-users] FS - MjSip no voice

[can_man at gmx.de]

Hello everyone,

I am trying to get FS working with the MjSip Java Sip-stack, the SipToSis source and the normal one. Everything works well within my own network and
when using x-lite, but when it comes to making calls from MjSip to an outside FS server I don't hear any voice - seems to be a NAT problem or some kind of other MjSip problem. Registration works fine though and SIP messages get through ok, but non of the UDP RTP ones. Would be great if someone could advice me on how to do the setup correctly. 

The whole FS trace can be found here: http://pastebin.freeswitch.org/8029

The settings for MjSip are:

"via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090",
"transport_protocols=udp tcp","from_url=<sip:puli at 91.101.58.142:5090>",
"username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes",
"#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068",
"audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500",
"bin_rat=rat","bin_vic=vic"


Thank you very much.
Best wishes,
Phil

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