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

[Alexandru Covalschi]

Hello,

I didn't follow your previous tread, but I suppose you use Kamailio as
'frontdoors' gate and route all your calls to external network using
Kamailio. In that way you'd better use RTPEngine (
https://github.com/sipwise/rtpengine) installed on Kamailio machine running
with external/internal interfaces.

Easiest thing is to put inside LOCATION route something like that

if route(FROMASTERISK) {
          rtpengine_manage(force trust-address direction=internal
direction=external);
}
else {
          rtpengine_manage(force trust-address direction=external
direction=internal);
}

(in route[FROMASTERISK] put a check to be sure call is comeing from your
asterisk)

and also - yes, define WITH_NAT if you're using standart configuration



2016-07-27 22:40 GMT+03:00 Tickling Contest <tickling.contest at gmail.com>:

> I added the #!define WITH_NAT option, and now the call can only be made
> one way. RTPProxy was started like so:
>
> $ rtpproxy -l 192.168.1.101 -s udp:localhost:7722 -u kamailio
>
> root at kamailioA:~# netstat -pln | egrep "kamailio|rtpproxy"
> tcp        0      0 192.168.1.101:5060      0.0.0.0:*
> LISTEN      10112/kamailio
> tcp        0      0 127.0.0.1:5060          0.0.0.0:*
> LISTEN      10112/kamailio
> udp        0      0 192.168.1.101:5060      0.0.0.0:*
>       10081/kamailio
> udp        0      0 127.0.0.1:5060          0.0.0.0:*
>       10081/kamailio
> udp        0      0 127.0.0.1:7722          0.0.0.0:*
>       10042/rtpproxy
> raw        0      0 0.0.0.0:255             0.0.0.0:*               7
>       10081/kamailio
> unix  2      [ ACC ]     STREAM     LISTENING     33357    10102/kamailio
>      /var/run/kamailio//kamailio_ctl
>
> My full config is at
> https://gist.github.com/ticklingcontest/e315972c80c82f6dfa23920c7725d60b
>
> BTW, my entire setup, kamailio, asterisk and the phones etc. are in one
> private network. I think setting realtime endpoint with "direct_media=no"
> is pointless as all of these interactions are fronted by Kamailio.
>
> What's going on here?  Any help is appreciated.
>
> On Wed, Jul 27, 2016 at 10:15 AM, Daniel Tryba <d.tryba at pocos.nl> wrote:
>
>> On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo wrote:
>> > 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.
>>
>> I'd suggest to use rtpengine for all calls, it fixes most problems and
>> uses nearly no resources (with the kernel plugin)
>>
>>
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>>
>
>
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>


-- 
Alexandru Covalschi
VoIP engineer and system administrator
tel: +37367398493
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