[Freeswitch-users] RTP problem/NAT

[Miha]

Hi Brian,

i have installed 1.4.12 on test environment (in production i have 1.4.8) 
and it does not send UPDATE in the middle of the call. The same UAC, 
router, server, same opensips.

What could be a reason that Fs will send UPDATE with sdp in the middle 
of the call? I need get this UPDATE wit sdp by FS, so that I will be 
able to fix this on opensips in production :)

But anyway I do not know why FS sends UPDATE with sdp as in 200 ok, 
which was send at the beginning of the call are the same sdp values as 
in UPDATE.

tnx
miha


On 17/10/2014 07:16, Miha wrote:
> 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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