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()).
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<br />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.
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<br />This tutorial for Mediaproxy is pretty old but should help you get started
<br />http://voiprookie.blogspot.com/2009/04/blog-post.html
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<br />On , Nick Khamis <symack@gmail.com> wrote:
<br />> Hey Duane,
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<br />> Thank you so much for your response. That is exactly my problem.
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<br />> Currently I only have
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<br />> OpenSIPS flowing SIP packets.
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<br />> As for the actual RTP, I was thinking of using either the STUN or
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<br />> NATHELPER module.
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<br />> Only I am not worried about NAT, I only need the flowing of RTP along
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<br />> with SIP. What
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<br />> is the most lightweight, and elgant solution to flow RTP? RTP Proxy
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<br />> from b2bua.org?
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<br />> I read somwhere that OpenSIPS also has a network hiding, and b2bua
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<br />> layer, is this
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<br />> the silver bullit I am looking for?
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<br />> Thanks in Advnace,
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<br />> Nick
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<br />> On Wed, Nov 30, 2011 at 5:08 PM, duane.larson@gmail.com> wrote:
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<br />> > Are your diagrams the path that SIP takes or RTP? Asterisk can be internal
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<br />> > and private from the outside world when it comes to SIP. It can also be
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<br />> > internal only when it comes to RTP but you would need to use a relay server
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<br />> > like Mediaproxy or RTPProxy. Mediaproxy can sit on the internet with a
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<br />> > public IP and Asterisk can be behind a firewall.
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<br />> > On , Nick Khamis symack@gmail.com> wrote:
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<br />> >> Hello Everyone,
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<br />> >> We are trying to close the doors entirely to our asterisk servers, making
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<br />> >> only opensips visible to the outside world:
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<br />> >> Incoming -> OpenSIPS -> Asterisk -> OpenSIPS -> Trunk
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<br />> >> | In Out
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<br />> >> | _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ |
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<br />> >> What I think we currently have is:
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<br />> >> Incoming -> OpenSIPS -> Asterisk -> OpenSIPS -> Trunk
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<br />> >> In | Out
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<br />> >> | _ _ _ _ _ _ _ _ _ __ _ _ |
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<br />> >> Without any port forwarding to the OpenSIPS box, everything
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<br />> >> works fine. With port forwarding, I get no audio both ways.
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<br />> >> If I am not mistaken, my questions are:
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<br />> >> * Can this be achieved
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<br />> >> * Do we have an externip, and port range settings for OpenSIPS.
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<br />> >> Thanks in Advnace,
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<br />> >> Nick
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<br />> >> _______________________________________________
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<br />> >>
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<br />> >> Users mailing list
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<br />> >> Users@lists.opensips.org
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<br />> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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<br />> >>
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<br />> >>
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<br />> > _______________________________________________
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<br />> > Users mailing list
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<br />> > Users@lists.opensips.org
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<br />> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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<br />> >
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<br />> _______________________________________________
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<br />> Users mailing list
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<br />> Users@lists.opensips.org
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<br />> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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