I have three OpenSIPS boxes running on Amazon EC2, and all working great (other than a couple issues I believe are related more to what I'm trying to do). I had to do a number of things to get it working, first is:<br>
<br>advertised_address="your public ip"<br><br>I recommend using elastic IPs<br><br>I then put record_route_preset in the INVITE section of the has_totag block in the config and in this block:<br><br> if (!is_method("REGISTER|MESSAGE")) {<br>
record_route_preset("public ip address");<br> }<br><br>I then use rtpproxy_offer and rtpproxy_answer in my route 1 blocks. Along with fix_nated_contact() if devices are registering from behind NAT (but that would be true of any install, not specific to EC2).<br>
<br>And then the trick, we had to patch rtpproxy to allow an -A parameter which is the advertise address (public IP). I submitted this to the developers, but I don't know if they updated the publicly available source yet.<br>
<br>Now to be clear, we do not pass through to a backend system like Asterisk yet (but are working on integrating with FreeSWITCH shortly). Our lines are registering directly to OpenSIPS.<br> <br>-dg<br>
<br><br><div class="gmail_quote">On Tue, Apr 27, 2010 at 1:47 PM, Brett Nemeroff <span dir="ltr"><<a href="mailto:brett@nemeroff.com">brett@nemeroff.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hello Brendan,<br>I'm doing a test of OpenSIPs in EC2 as well.. All is
working with the exception of ACK received for 500 type errors which are
being retransmitted. Error logs seem to indicate that the dialog should
have been destroyed and I'm not sure why I'm getting the retrans. I
posted this question a few weeks ago.. <br>
<br>I didn't try record_route_preset, but I did try
"advertised_address", which may help. <br><br>I'll let you know if I get
a resolution on my side. <br><div><div></div><div class="h5"><br><div class="gmail_quote">On Tue, Apr 27, 2010 at 3:39 PM, Brendan Sterne <span dir="ltr"><<a href="mailto:brendan@callvine.com" target="_blank">brendan@callvine.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Greetings,<br>
<br>
I am experimenting with using OpenSIPS in Amazon EC2 to distribute<br>
calls to Asterisk instances (also running in Amazon EC2). The<br>
challenge is that servers on Amazon EC2 have private IPs to<br>
communicate with each other, but different public IPs when accessed<br>
from without EC2. Basically Amazon has a Static NAT setup that does<br>
IP translation (but not port translation). Amazon provides a public<br>
DNS name that resolves to the public host IP outside of Amazon EC2,<br>
and to the private host IP inside of Amazon EC2.<br>
<br>
I know that it is not recommended to use OpenSIPS behind a NAT, but<br>
I'm curious if I can make this work. Right now I'm focusing on<br>
inbound calls, SIP call control only (I will use nathelper / rtpproxy<br>
as necessary to help with media later).<br>
<br>
Here's the scenario for the invite:<br>
Soft Phone -------> EC2 Firewall -----> OpenSIPS -----> Asterisk<br>
<br>
The INVITE,100,200 works fine - I have opensips redirect to the<br>
Asterisk using<br>
rewritehostport(<asterisk_dns_name>);<br>
And I use record_route_preset() to record the Public DNS in the<br>
route. This will create a Route Set that will work both ways (from<br>
the Soft Phone, and from the Asterisk).<br>
record_route_preset(<opensips_dns_name>)<br>
<br>
The problem I'm having is with the ACK. It is being routed from the<br>
Soft Phone to the OpenSIPS via it's Amazon DNS name (<a href="http://ec2-204-236-245-16.compute-1.amazonaws.com" target="_blank">ec2-204-236-245-16.compute-1.amazonaws.com</a><br>
), but the OpenSIPS isn't recognizing the name as a local alias. I<br>
have alias set: alias="<a href="http://ec2-204-236-245-16.compute-1.amazonaws.com" target="_blank">ec2-204-236-245-16.compute-1.amazonaws.com</a>:<br>
5060", but the opensips log shows "Topmost URI is NOT myself" (you can<br>
see more below). I have attached my config, logs, and a sip trace.<br>
<br>
Any suggestions are appreciated.<br><br></blockquote></div>
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