Re: [GnomeMeeting-devel-list] SIP, RTP ports and firewall



Damien Sandras a écrit :

Le lundi 21 novembre 2005 à 23:08 +0100, Daniel Huhardeaux a écrit :
Damien Sandras a écrit :

[...]

Call didn't pass. "Fin anormale de l'appel"
Right. See this :
2005/11/21 19:58:55.364 0:59.816 SIP Handler:8499268 RTP STUN could not create socket pair!

Are you 100% sure you have run the NAT druid test and that it reported
something different than "symmetric NAT" ?



Damien,

as told at the begining of this discussion, if I go back with gconf-editor to the RTP ports 6170:7170 or 6170:6190 it's ok, just have then the problem of only one or two call able in a row.

I know that, my question stays valid...
I started gconf-editor and modify RTP ports to 6970:6990, then run GM, then the druid: "NAT avec restriction de port" (NAT with restriction port), called 612 fwd pulver com and call went smooth.

I closed GM, modify RTP ports to 16384:16388, run the druid: "NAT avec restriction de port" (NAT with restriction port), called 612 fwd pulver com and call failed like reported above.

I have some difficulties to understand why changing the RTP ports should interfere with STUN. Remember that registration is always ok.
--
Daniel



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