[OpenSIPS-Users] NAT traversal issue in case of different internal and external ports

[Krunal Patel]

Hi

All this issue are fixed.
I have put force_rport in main route.
That has fixed signalling issue in case of different ports.

Thanks,
Krunal Patel

On Tue, Dec 16, 2008 at 8:04 PM, Krunal Patel <krunal.lists at gmail.com>wrote:

> Hi
>
> Sorry but please ignore last post.
> At present I have put force_rport in main route.
>
> I have registered 4 UAC using same username .
> Among them 2 are on public ip & the other 2 are behind nat.
> now when I am calling to PSTN signalling & audio both works properly.
>
> But when getting incoming call from PSTN & pickup call using the UAC which
> is behind NAT I am getting NO audio.
>
> It works if I register single UAC using a username.
> I am having an issue when I do fork.
>
> Please advice what can be the issue.
>
> Thanks
> Krunal Patel
>
>
> On Tue, Dec 16, 2008 at 7:36 PM, Krunal Patel <krunal.lists at gmail.com>wrote:
>
>> Hi,
>>
>> I am doing force_rport.
>> & that's why external port is being added in via
>> Via: SIP/2.0/UDP
>> [INTERNAL_IP]:5065;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-a85eb9f3;rport=50034.
>>
>> But from reply route, reply is not being sent to correct port which is in
>> via.
>> Please have a look to below given 183 progress trace
>>
>> U YYY.YYY.YYY.YYY:5060 -> XXX.XXX.XXX.XXX:5065
>> SIP/2.0 183 Session Progress.
>> Call-ID: df9cc79e-ffa2ffb3@[INTERNAL_IP].
>> CSeq: 102 INVITE.
>> From: "FROM_NUM" <sip:FROM_NUM at voip.convergenze.it<sip%3AFROM_NUM at voip.convergenze.it>
>> >;tag=45a59ea2bc86747fo0.
>> To: <sip:TO_NUM at voip.convergenze.it <sip%3ATO_NUM at voip.convergenze.it>
>> >;tag=3eddfbdb7400749.
>> Via: SIP/2.0/UDP
>> [INTERNAL_IP]:5065;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-a85eb9f3;rport=50034.
>>
>> Let me know hwat can be the issue?
>>
>> On Tue, Dec 16, 2008 at 5:39 PM, Krunal Patel <krunal.lists at gmail.com>wrote:
>>
>>> Hi
>>> I put force_rport at the beginning of main route.
>>> now signaling issue is fixed but facing audio issue.
>>> & getting error: empty response from mediaproxy for PSTN to behind nated
>>> users.
>>>
>>>
>>>
>>> On Mon, Dec 15, 2008 at 7:58 PM, Jeff Pyle <jpyle at fidelityvoice.com>wrote:
>>>
>>>>  Iñaki,
>>>>
>>>> Some time back I had a lab setup that used force_rport() on everything
>>>> that came in.  This caused problems if the INVITE came from a Broadworks
>>>> server.  Broadworks does not use source port 5060 on its traffic, for load
>>>> handling they say.  Anyway, my replies went to the source port Broadworks
>>>> used instead of to 5060.  And, of course, failed.
>>>>
>>>> I never did spend time on how to handle this properly.  Perhaps someone
>>>> here has some thoughts.
>>>>
>>>>
>>>> - Jeff
>>>>
>>>>
>>>>
>>>>
>>>> On 12/15/08 8:05 AM, "Iñaki Baz Castillo" <ibc at aliax.net> wrote:
>>>>
>>>> > El Lunes, 15 de Diciembre de 2008, Krunal Patel escribió: >
>>>> > force_rport(); Use force_rport(); at the beginning of your script,
>>>> just
>>>> > there. And take if off from on_reply_route. -- Iñaki Baz Castillo
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>
>
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