[Freeswitch-users] FS uses private rtp address in the SDP

[afshin afzali]

Hi Miguel,

I do not understand how the ACL (if you mean apply-nat-acl) can effects on
using public IP in SDP when I specified both ext-sip-ip and ext-rtp-ip in
sip profile!
I changed the apply-nat-acl to a "always allow" and "rfc1918" but does not
make difference.

Afshin


On Thu, Aug 14, 2014 at 3:18 PM, Miguel Oyarzo <miguelaustro at gmail.com>
wrote:

>
> Wrong ACL?
>
>
> On Wed, Aug 13, 2014 at 10:06 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
>> Hi FreeSWITCHers:
>>
>> I configured a sip profile as bellow:
>>
>>
>> =================================================================================================
>> Name                    sipinterface_1
>> Domain Name             N/A
>> Auto-NAT                false
>> DBName                  sofia_reg_sipinterface_1
>> Pres Hosts
>> Dialplan                XML
>> Context                 context_1
>> Challenge Realm         auto_to
>> RTP-IP                  192.168.10.59
>> Ext-RTP-IP              X.Y.Z.W
>> SIP-IP                  192.168.10.59
>> Ext-SIP-IP               X.Y.Z.W
>> URL                     sip:mod_sofia at X.Y.Z.W:5060
>> BIND-URL                sip:mod_sofia at X.Y.Z.W
>> :5060;maddr=192.168.10.59;transport=udp,tcp
>> HOLD-MUSIC              N/A
>> OUTBOUND-PROXY          N/A
>> CODECS IN               G722,PCMU,PCMA,GSM
>> CODECS OUT              G722,PCMU,PCMA,GSM
>> TEL-EVENT               101
>> DTMF-MODE               rfc2833
>> CNG                     13
>> SESSION-TO              0
>> MAX-DIALOG              0
>> NOMEDIA                 false
>> LATE-NEG                false
>> PROXY-MEDIA             false
>> ZRTP-PASSTHRU           false
>> AGGRESSIVENAT           true
>> CALLS-IN                0
>> FAILED-CALLS-IN         0
>> CALLS-OUT               0
>> FAILED-CALLS-OUT        0
>> REGISTRATIONS           1
>>
>> for every INVITE that comes in via this profile, FS replys with this SDP:
>>
>>    v=0
>>    o=FreeSWITCH 1407905843 1407905844 IN IP4 192.168.10.59
>>    s=FreeSWITCH
>>    c=IN IP4 192.168.10.59
>>    t=0 0
>>    m=audio 25142 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=ptime:20
>>
>> Although the Contact header of 200 OK is correct :
>> Contact: <sip:4000 at X.Y.Z.W:5060;transport=udp>
>>
>>
>> Appreciate all comments,
>> -- Afshin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
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>>
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>>
>>
>
>
> --
> "Efficiency is doing things right; effectiveness is doing the right things
> (Peter Drucker)"
>
> Miguel Oyarzo
> DevOps & VoIP Engineer
> Linux User: # 483188 - counter.li.org
> http://au.linkedin.com/in/mikeaustralia
> Melbourne, Australia
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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