[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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