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.cont...@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@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.tr...@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)
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> _______________________________________________
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>
>
--
Alexandru Covalschi
VoIP engineer and system administrator
tel: +37367398493
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