First, this process should happen quicker. If it doesn't, it's probably because FS does not receive any RTP traffic from your phone. Perhaps your cellular network is doing "things" in order to prevent VoIP. FS can't use the external IP from the start. FS has no way to know the exact port it should use to reach a specific device behind NAT, because until this device sends RTP, the port does not exist yet. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 06/09/2010 à 02:59, Ross McKillop a écrit :
Hi,
I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic.
No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console;
2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263
After which the call works perfectly.
Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start?
Any assistance you can offer would be much appreciated.
Best Regards, Ross _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org