[Freeswitch-users] RTP problem/NAT

[Chris Tunbridge]

did you turn on aggressive-nat-detection inside of
sip_profiles/internal.xml what about NDLB, i have a box running that exact
version without any issues regarding NAT

On Thu, Oct 16, 2014 at 3:40 AM, Miha <miha at softnet.si> wrote:

> Hi,
>
> first my FS is on public ip.
>
> We have migrated  from 1.2.x to Version
> 1.4.8+git~20140910T181155Z~9f72b11892~64bit (git 9f72b11 2014-09-10
> 18:11:55Z 64bit as we needed different server.
>
> I have noticed that a lot of costumers are complaining that a lot of
> phone call has been cut in about 1:30 min of the call. From what I can
> see from sip trace is that FS begins to send media to private ip.
>
> link to trace: http://pastebin.com/9M6X6SFp
>
> So, rtp is send first to right ip and media is ok but in about 1:30 min
> of the call the FS sends update and begin to send rtp to private ip.
>
>
> thx for help!
>
> miha
>
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