[Freeswitch-users] RTP problem/NAT

[Miha]

Hi Brian,

i think i found an issue. Please let me know if aggressive-nat-detection 
will help as a do not find any appropriate explenation what exactly does 
it do.

Freeswitch in my case is for RTP, so opensips relays all call throught 
FS and than FS sends it back to opensips which sends a call to UAC.

Problem is that Fs sends in the middle of the call UPDATE request (which 
is ok) and than UAC sends back 200 ok. In this 200 ok the contact 
address is private (also in SDP) as opensips does not replace contact ip 
with public source ip , so FS begins to send media to private ip which 
is the reason why media stops.

I think this problem was also present with 1.2.x version but FS did not 
sends updates. Do not understand this sentence wrong as FS is not 
behaving wrong i am just explaning why I have this in new version :)

Will Aggressive nat detection fix this or I should fix this on opensips 
and what exactly does aggresive nat detection do?

tnx!

miha

On 16/10/2014 22:14, Chris Tunbridge wrote:
> 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 
> <mailto: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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