I have defined the rtp port range for 12000-12100 in switch.conf.xml. However Freeswitch is offering a port 48320 in the invite message. The result is, that the incoming RTP stream is blocked by the firewall (I can see a reject for UDP 48320). Any hint how to solve this? Best regards Peter See config and invite message: <!--RTP port range --> <param name="rtp-start-port" value="12000"/>--> <param name="rtp-end-port" value="12100"/>--> Invite: INVITE sip:027xxxxxxxx@sip.itsp.de SIP/2.0. Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm. Max-Forwards: 69. From: "0608xxxxxxx" <sip:4918xxxxxxxxxxxx@sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp. To: <sip:027xxxxxxxx@sip.itsp.de>. Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59. CSeq: 126174137 INVITE. Contact: <sip:gw+tgnet3@217.24.xx.xxx:5080;transport=udp;gw=tgnet3>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. P-Key-Flags: keys="3". X-FS-Support: update_display. Remote-Party-ID: "0608xxxxxxx" <sip:0608xxxxxxx@sip.itsp.de>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx. s=FreeSWITCH. c=IN IP4 217.24.xx.xxx. t=0 0. m=audio 48320 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20.