[SR-Users] bypass rtp traffic.

Iñaki Baz Castillo ibc at aliax.net
Sat Jul 16 11:03:08 CEST 2011


Hi. Kamailio and rtpproxy does NOT decide to send rtp to asterisk. It is
asterisk who decides to receive it and that entirely  depends on asterisk
sip condigurarion and asterisk sip peers configuration.

Your question is not related to kamailio but just to asterisk.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110716/37f46fb2/attachment-0001.htm>


More information about the sr-users mailing list