Hello,

I have my setup at Norway and some of my customers are in USA so i have concern about audio latency.

I want the configuration such a way that if USA customer is calling than the rtp should route through my USA freeswitch servers however sip should route through Norway setup.

Is there any way i can achieve this?

I have tried modifying rtp-ip but i am getting 'DESTINATION_OUT_OF_ORDER'

2017-05-02 08:50:42.650711 [ERR] switch_core_media.c:5843 AUDIO RTP REPORTS ERROR: [Bind Error! 65.11.17.9:25146]
2017-05-02 08:50:42.650711 [NOTICE] switch_core_media.c:5844 Hangup sofia/default/12345679 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]