You have proxy media on don't you? From what this looks like you have an inbound invite from a Snom and we have an outbound invite. If you're doing something such as proxy media it will try to pass it thru as is... Giving you little or NO control over the port. Unless you fix your endpoint to also use the same restrictive port range. Because by default we will not use anything in the 48000 range .. the giveaway here is the fact I see P-Key-Flags: header which is indication you're using a snom with proxy media. Am I correct? /b On Jan 27, 2010, at 4:14 AM, Peter P GMX wrote:
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.
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