<div dir="ltr">Hi Fred<br><div><br></div><div>So i am running rtpengine but prob i am having is i receive the call as rtp and need it to become srtp before i send it to the carrier</div><div>then the carrier sends me a call as srtp and need that to become normal rtp before i can go to the uac</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Oct 14, 2020 at 3:27 PM Fred Posner <<a href="mailto:fred@palner.com">fred@palner.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">On Wed, 2020-10-14 at 11:51 +0100, Andy Kama wrote:<br>
> Hi All,<br>
> <br>
> I have setup a tls connection to a sip provider and it works<br>
> perfectly<br>
> But they are requesting i send srtp<br>
> i can find any documentation how i do it with rtpengine etc?<br>
> <br>
> any guidance would be appreciated<br>
> <br>
> <br>
<br>
Generally this would be initiated by the endpoint making/receiving the<br>
call and can be as simple as making sure it's offered in the phone<br>
settings.<br>
<br>
If you need to bridge rtp with srtp, you can look at rtpengine (<br>
<a href="https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html" rel="noreferrer" target="_blank">https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html</a>)<br>
for information on flagging the offer/manage.<br>
<br>
Of course, this would also require you to install and run rtpengine.<br>
<br>
-- <br>
Fred Posner<br>
<a href="mailto:fred@palner.com" target="_blank">fred@palner.com</a><br>
<a href="https://www.palner.com" rel="noreferrer" target="_blank">https://www.palner.com</a><br>
<br>
Need Fred? Call Fred. 336-HEY-FRED<br>
Matrix: @fred:<a href="http://matrix.lod.com" rel="noreferrer" target="_blank">matrix.lod.com</a><br>
<br>
<br>
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</blockquote></div>