[SR-Users] bypass rtp traffic.

[Klaus Darilion]

I guess you forward all calls via Asterisk.

Yes: set canreinvite=yes (name was changed in newer Asterisk versions)
in sip.conf for the peers and Asterisk will send reINVITEs after call
setup to offload RTP.

regards
Klaus

Am 13.07.2011 07:43, schrieb MingHon:
> Hi List,
> 
> i would like to know is it possible to bypass the rtp traffic forwarding
> to asterisk server?
> 
> my kamailio and rtpproxy is on the same box and asterisk is on the other
> box.
> 
> can kamailio/rtpproxy handle the rtp traffic without forwarding to
> asterisk box?
> 
> thanks in advance.
> 
> 
> -- 
> Regards,
> 
> MingHon
> 
> 
> 
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