<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>Doesn't this cause more problems? I thought bypass media behind NAT or in mixed systems - NAT/no NAT - is a mess.</div><div><br></div><div>I would love to have a local (dumb) proxy on each site to handle the local calls but all the intelligence (user database/ management) should remain in the centralized master.</div><div><br></div><div><br></div><div><br></div><div><br><br><br></div><div><br>Am 24.04.2015 um 17:36 schrieb Russell Treleaven <<a href="mailto:rtreleaven@bunnykick.ca">rtreleaven@bunnykick.ca</a>>:<br><br></div><blockquote type="cite"><div><div dir="ltr">As an alternative you could try bypass media.<div><a href="https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview">https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview</a><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Apr 24, 2015 at 11:17 AM, Roman Dissauer <span dir="ltr"><<a href="mailto:roman@dissauer.net" target="_blank">roman@dissauer.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Dear all,<br>
<br>
I’m running a centralized Freeswitch based PBX for use on several sites. All phones register against this Freeswitch instance.<br>
<br>
Now I want to install SIP and RTP Proxies on every site to keep RTP traffic locally on site. Registration should still be done by freeswitch. Can anybody give me a hint if this is possible with opensips and rtpproxy?<br>
<br>
Maybe I can clarify it a bit more:<br>
----------------<br>
| Freeswitch |<br>
----------------<br>
|<br>
| Public Internet<br>
|<br>
|--------- Phone 3 external<br>
|<br>
|<br>
------------------<br>
| Firewall / NAT |<br>
------------------<br>
|<br>
----------------<br>
| Proxy | Site 1<br>
----------------<br>
|<br>
|<br>
|--------- Phone 1 internal<br>
|--------- Phone 2 internal<br>
<br>
Phone 1 - 3 are all registered at Freeswitch<br>
Phone 1 calls Phone 3: SIP and RTP over Freeswitch<br>
Phone 1 calls Phone 2: SIP over Freeswitch but RTP over Proxy<br>
<br>
Does this make sense?<br>
<br>
Thanks,<br>
Roman<br>
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</blockquote></div><br></div>
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