[Freeswitch-users] SIP provider with extern rtp server

[Ivan C Myrvold]

The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the  
183 Session Progress message on the server, but just in one direction,  
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the  sip  
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

> Make sure you let their media IPs through your firewall. Also, if you
> are behind a NAT, check you have things passing to the correct
> internal address.
>
> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> I have used a SIP provider for more than a year. A few days ago, he
>> said he was moving to a new server, and asked me to reconfigure. I
>> did, and everything seemed to work fine, until I did an outgoing call
>> to an external telephone. I found out I had no audio, in neither
>> direction. Incoming calls was working fine.
>>
>> My provider said that the rtp is not going through the sip server, as
>> it did earlier, but now through several other IP's.
>>
>> Do I have to do some special configuration to handle that?
>>
>> Ivan
>>
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>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
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> live; not live to eat.) ~Marcus Tullius Cicero
>
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