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

[Jurijs Ivolga]

Hi Tickling,

Could you please share with us your working config?

With kind regards,

Jurijs

On Wed, Jul 27, 2016 at 8:54 AM, SamyGo <govoiper at gmail.com> wrote:

> 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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>>
>
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