[SR-Users] Kamailio LB: how to get Asterisk to do the RTP

[SamyGo]

Hi Again,

You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then
depending upon how your clients will interact with asterisk you may or may
not need a media proxy, like RTPproxy. If asterisks can send/receive media
directly from the internet then its ok for now, else you definitely need to
have rtpproxy/rtpengine in there.


Regards,
Sammy


On Tue, Jul 26, 2016 at 10:29 PM, Tickling Contest <
tickling.contest at gmail.com> wrote:

> With the help of members from this mailing list (many thanks!), I finally
> got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make
> a call using the setup that looks like this:
>
> [Kamailio 4.4.2]<->[Asterisk 13.7.2]
>
> Kamailio manages REGISTERs, but also forwarding them to Asterisk.
>
> I am able to make a call, but I get only one way audio or no audio
> depending on which client made the call (SipDroid->Zoiper I hear one way
> audio on Zoiper, but no audio if the call is made the other way). I noticed
> that Kamailio forced direct media between the endpoints in this situation,
> but my application really needs Asterisk to handle it.
>
> How do I do this? Should I start by forwarding INVITEs to Asterisk? How do
> I do that?
>
> Any help is appreciated.
>
> Thanks!
>
>
>
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