[Freeswitch-users] NAT Problem with UPNP only for RTP - SIP tcp keept open

[Paul Muaddib]

Hi Kevin,

Yes, Deutsche Telekom probably does some kind of ALG to make it "easier"
But why does not the echo application work and normal voice call does work?

Regards,
Paul

Am Di., 20. Nov. 2018 um 23:36 Uhr schrieb Kevin Olbrich <ko at sv01.de>:

> Sounds like your gateway does SIP ALG which in my experience breaks SIP in
> 99 percent of all communication (because it fixes where nothing is broken).
> Had such problems when I specified a STUN server for clients - NAT
> traversal should be left to the server alone. It's perfectly fine to use
> private IPs in SIP/SDN and let RPORT handle it all.
>
> Kevin
>
>
> Am Do., 15. Nov. 2018 um 23:51 Uhr schrieb Paul Muaddib <
> paul.muaddib83 at gmail.com>:
>
>> Hi
>>
>> at the moment I have the setup
>>
>> Internal.xml
>> ext-rtp-ip = auto-nat
>> ext-sip-ip = auto-nat
>>
>> External.xml
>> ext-rtp-ip = “my_local_freeswitch_ip”
>> ext-sip-ip = auto-nat
>>
>> RTP ports (only) are begin opened by UPNP and SIP via TCP is being keep
>> open with “expire-seconds=600” and “register=true” (Is there an alternative
>> for the expire-seconds ? nat-options-ping is only for endpoints registering
>> to freeswitch, right?)
>>
>> The reason why I am doing this is, that I dont want to open up a SIP Port
>> in the Firewall.
>>
>> But this setup up doesn’t seem right. Audio is working, inbound and
>> outbound, both ways. But the echo application is not.
>>
>> When I change
>> External.xml
>> ext-rtp-ip = “auto-nat”
>>
>> I get one-way audio for inbound calls for the callee. The caller can be
>> heard.
>>
>> I tried to analyses it in wireshark, but I don’t understand it. The sip
>> provider keeps sending INVITE messages while freeswitch is confirming it
>> with OK but keeps going on until the connection breaks.
>>
>> Thank you for your help
>>
>>
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