[SR-Users] bypass rtp traffic.

Klaus Darilion klaus.mailinglists at pernau.at
Thu Jul 21 09:48:27 CEST 2011


You should post a SIP trace, together with the IP addresses of all nodes:

ngrep -t -d any -P "" -Wbyline port 5060

If there is sensitive information in the traces, just remove/replace it.

regards
Klaus

Am 21.07.2011 09:23, schrieb MingHon:
> Hello List,
> 
> im still trying but no luck.
> asterisk canreinvite already set to yes
> 
> 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
> 
> 
> 
> _______________________________________________
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> sr-users at lists.sip-router.org
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