[OpenSIPS-Users] OpenSIPS as the Firewall

[duane.larson at gmail.com]

I use Mediaproxy and don't have any issues with it. I have no experience  
with RTPProxy. OpenSIPS b2bua is not what you are looking for when it comes  
to RTP. The B2B_LOGIC module can do network hiding (b2b_init_request("top  
hiding")) but so can the dialog module now (topology_hiding()).

With Mediaproxy you can just add more Mediaproxies if the first one is  
getting used too much. So from a loadbalancing and HA perspective  
Mediaproxy is very simple to deploy. It also supports ICE so that could be  
good depending on your clients.

This tutorial for Mediaproxy is pretty old but should help you get started
http://voiprookie.blogspot.com/2009/04/blog-post.html





On , Nick Khamis <symack at gmail.com> wrote:
> Hey Duane,





> Thank you so much for your response. That is exactly my problem.


> Currently I only have


> OpenSIPS flowing SIP packets.





> As for the actual RTP, I was thinking of using either the STUN or


> NATHELPER module.


> Only I am not worried about NAT, I only need the flowing of RTP along


> with SIP. What


> is the most lightweight, and elgant solution to flow RTP? RTP Proxy


> from b2bua.org?


> I read somwhere that OpenSIPS also has a network hiding, and b2bua


> layer, is this


> the silver bullit I am looking for?





> Thanks in Advnace,





> Nick








> On Wed, Nov 30, 2011 at 5:08 PM, duane.larson at gmail.com> wrote:


> > Are your diagrams the path that SIP takes or RTP? Asterisk can be  
> internal


> > and private from the outside world when it comes to SIP. It can also be


> > internal only when it comes to RTP but you would need to use a relay  
> server


> > like Mediaproxy or RTPProxy. Mediaproxy can sit on the internet with a


> > public IP and Asterisk can be behind a firewall.


> >


> >


> >


> >


> > On , Nick Khamis symack at gmail.com> wrote:


> >> Hello Everyone,


> >>


> >>


> >>


> >> We are trying to close the doors entirely to our asterisk servers,  
> making


> >>


> >> only opensips visible to the outside world:


> >>


> >>


> >>


> >> Incoming -> OpenSIPS -> Asterisk -> OpenSIPS -> Trunk


> >>


> >> | In Out


> >> |


> >>


> >> | _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ |


> >>


> >>


> >>


> >>


> >>


> >> What I think we currently have is:


> >>


> >>


> >>


> >> Incoming -> OpenSIPS -> Asterisk -> OpenSIPS -> Trunk


> >>


> >> In | Out


> >> |


> >>


> >> | _ _ _ _ _ _ _ _ _ __ _ _ |


> >>


> >>


> >>


> >> Without any port forwarding to the OpenSIPS box, everything


> >>


> >> works fine. With port forwarding, I get no audio both ways.


> >>


> >>


> >>


> >> If I am not mistaken, my questions are:


> >>


> >> * Can this be achieved


> >>


> >> * Do we have an externip, and port range settings for OpenSIPS.


> >>


> >>


> >>


> >> Thanks in Advnace,


> >>


> >>


> >>


> >> Nick


> >>


> >>


> >>


> >> _______________________________________________


> >>


> >> Users mailing list


> >>


> >> Users at lists.opensips.org


> >>


> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


> >>


> >>


> > _______________________________________________


> > Users mailing list


> > Users at lists.opensips.org


> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users


> >





> _______________________________________________


> Users mailing list


> Users at lists.opensips.org


> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20111201/222ac1ff/attachment-0001.htm>