<div dir="ltr">Hi,<div><br></div><div>which debug tool is that in your screenshot?</div><div><br></div><div>Thank you very much!</div><div><br></div><div>Kevin<br><br><br><div class="gmail_quote"><div dir="ltr">Am Mi., 22. Aug. 2018 um 18:47 Uhr schrieb Sergiu Pojoga <<a href="mailto:pojogas@gmail.com">pojogas@gmail.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Why would the UAC need to 'whitelist' Asterisk?<div><br></div><div>Here's an example of an inbound call from Asterisk to a UAC with a sip proxy in the middle.</div><div>Notice how the proxy mangles SDP in the 200 OK reply. Media will flow directly.</div><div><br></div><div>Not saying you shouldn't figure it out how to setup and use rtp proxies, just that it's not a must in all scenarios, especially when you have a media gateway such as Asterisk listening on a public interface.<br><div><br></div><div><div><img src="cid:ii_jl5d01ch0" alt="image.png" width="521" height="395"><br></div></div><div><br></div></div></div><br><div class="gmail_quote"><div dir="ltr">On Wed, Aug 22, 2018 at 12:21 PM Wilkins, Steve <<a href="mailto:swwilkins@mitre.org" target="_blank">swwilkins@mitre.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal">Hello,<u></u><u></u></p>
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<p class="MsoNormal">Yes, I don’t the UAC to have to whitelist the Asterisk Server. Everything works great for SIP traffic and I want the same thing for RTP traffic, but I just can’t get rtpengine to work (and I have tried many configurations).<u></u><u></u></p>
<p class="MsoNormal"><b>From:</b> sr-users <<a href="mailto:sr-users-bounces@lists.kamailio.org" target="_blank">sr-users-bounces@lists.kamailio.org</a>> <b>
On Behalf Of </b>Sergiu Pojoga<br>
<b>Sent:</b> Wednesday, August 22, 2018 11:23 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Struggling with RTPProxy and RTPEngine<u></u><u></u></p>
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<p class="MsoNormal">Is there a particular reason why you'd want to involve an rtp proxy in your setup? <u></u><u></u></p>
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<p class="MsoNormal">Considering Asterisk is facing a public interface, media can be set up to flow direct between UACs and Asterisk.<u></u><u></u></p>
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<p class="MsoNormal">On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba <<a href="mailto:d.tryba@pocos.nl" target="_blank">d.tryba@pocos.nl</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal">On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:<br>
> My offer and answer =><br>
> rtpengine_offer("trust-address replace-session-connection replace-origin");<br>
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If this is really your config you should change the offer to answer for<br>
the answer part of the config. Either use rtpengine_offer and<br>
rtengine_answer in the correct places of simply use rptengine_manage and<br>
let the module figure out the right command.<br>
<br>
I.O.W. you aren't giving any indepth info on your config.<br>
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