14 Jul
2011
14 Jul
'11
4:09 p.m.
Hi,
yup i tried "canreinivte=yes" in sip.conf and also in the extension database.
urm how bout having direct rtp traffic and also relay rtp traffic in my setup?
example, UA1 and UA2 is at the same nat. so UA1 and UA2 will have direct rtp traffic.
UA1 <--(rtp)--> UA2.
and UA3 and UA4 both behind different nat will need relay rtp traffic. when invite compare the "received:ip_address".
UA3 <--(rtp)--> KAMAILIO <--(rtp)--> UA4
issit possible to have both? urm ya what is the variable for the "received:ip_address" ?
Thanks.
--
Regards,
MingHon
On Thu, Jul 14, 2011 at 11:22 PM, Carsten Bock carsten@ng-voice.com wrote:
> Hi,
>
> have you tried "canreinvite=yes" on your Asterisk-box?
> If that does not help, there is probably no way to make the
> RTP-Traffic bypass your asterisk box...
>
> Carsten
>
>
> 2011/7/14 MingHon gminghon@gmail.com:
> > Hello,
> > anyone?
> > currently my setup look like this.
> > when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk.
> > [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2]
> > ^
> > |
> > RTP TRAFFIC
> > |
> > v
> > [ASTERISK]
> > what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio
> but
> > not to asterisk.
> > can kamailio handle the rtp traffic it own?
> > [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2]
> > ^
> > |
> > X
> > |
> > v
> > [ASTERISK]
> >
> > Thanks in advance.
> > --
> > Regards,
> >
> > MingHon
> >
>
>
>
> --
> Carsten Bock
> http://www.ng-voice.com
> mailto:carsten@ng-voice.com
>
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