[SR-Users] bypass rtp traffic.

Skyler skchopperguy at gmail.com
Tue Jul 26 14:02:18 CEST 2011


Hi,

On Thu, 2011-07-21 at 15:23 +0800, MingHon wrote:
> Hello List,
> 
> 
> im still trying but no luck.
> asterisk canreinvite already set to yes
> 

 What version of asterisk? I think in 1.6.2 canreinvite was replaced
with directmedia and directrtp.

> 
> now im testing in lan
> i setup kamailio and asterisk in same lan 
> kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23
> 
> 
> canreinvite=yes in asterisk. when both ua in the same lan 
> register directly to asterisk the reinvite work. both ua will have
> and direct media flow
> 
> 
> [ua1]<====>[ua2]
>              |
>              |
>              x
>              |
>              v
>        [asterisk]
> 
> 
> when ua register to kamailio the audio work and the reinvite message
> is same as the first invite message.
> 
> 
> [ua1]<====>[kamailio]<====>[ua2]
>                         |  ^
>                         |  |
>                         |  |
>                         v |
>                    [asterisk]
> 
> 
> how do i stop the media flow between kamailio and asterisk? 
> make kamailio relay the rtp between both ua.
> 
> 
> [ua1]<====>[kamailio]<====>[ua2]
>                         |  ^
>                         x x
>                         |  |
>                         v |
>                    [asterisk]
> 
> 
> 
> 
> anyone could give some hint?
> 
> 
> thanks in adv.
> 
> 
> -- 
> Regards,
> 
> MingHon

S.





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