Re: [SR-Users] bypass rtp traffic.

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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