[OpenSIPS-Users] Relaying of RTP packets between two opensips/mediaproxy

[Adrian Georgescu]

MediaProxy is designed to work only on public IP addresses. Only the  
end-points can or may have private IP addresses. If your server has a  
private IP address it cannot detect if the end-points are behind NAT  
or not.

As a general rule servers must run on a public IP address otherwise  
you will sooner or later run into problems that are very expensive to  
solve.

Adrian

On Apr 28, 2010, at 5:00 PM, José María Jiménez wrote:

> Hi all,
>
> I'm testing a VoIP architecture in order to make a call between two  
> IP phones in a LOCAL network environment (I'm not using any public  
> IP) where the packets through 2 proxies (composed by a OpenSIPS, a  
> MediaProxy module and a MediaProxy) and an Asterisk. After several  
> attempts of configuring OpenSIPS and MediaProxy, I can't achieve RTP  
> relaying between P1, P2 and Asterisk.
>
> VoIP Architecture:
>
>            192.x.x.x                172.x.x.x                172.y.x.x
> +------------------------------+--------------------------- 
> +----------------------------+
>
> S1-+
>       |
>       +<---SIP/RTP---> P1 <---SIP/RTP---> P2 <---SIP/RTP---> A
>       |
> S2 -+
>
>
>    S1 = Softphone 1
>    S2 = Softphone 2
>
>    P1 = Proxy 1 ( OpenSIPS + MediaProxy )
>    P2 = Proxy 2 ( OpenSIPS + MediaProxy )
>
>    A = Asterisk
>
> SIP traffic works correctly:
> - The phones are registred (REGISTER) in Asterisk. OpenSIPS 1 and 2  
> only relay the SIP packets.
> - The caller sends the initial "INVITE" to P1, P1 to P2 and P2 to  
> Asterisk.
>
> When I make a call, signaling works correctly but audio (RTP)  
> doesn't. The phones send their RTP packets to Proxy1, But P1 is  
> unable to forward them to P2. I know it can be due to NAT problems,  
> but I still have some doubts:
>
> 1) About the NAT problem, Does it affect to local networks? All  
> elements of this architecture are in differents local networks  
> (phones in 192.x.x.x, proxies in 172.x.x.x, asterisk in 172.y.x.x)  
> and every element of the solution knows how to routing IP packets  
> from one network to another, so... is NAT affecting to this VoIP  
> architecture?
>
> 2) Another question... It's possible relaying RTP traffic from one  
> MediaProxy directly to another, right?
>
> Thanks in advance for your help :) I have read a lot about NAT but I  
> still don't understand if this affects to my VoIP architecture if I  
> work just with private IPs.
>
> José M.
>
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