[Freeswitch-users] Wrong RTP port submitted?

[Peter P GMX]

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 at 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 at sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp.
To: <sip:027xxxxxxxx at sip.itsp.de>.
Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59.
CSeq: 126174137 INVITE.
Contact: <sip:gw+tgnet3 at 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 at 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.