[SR-Users] rtpproxy and kamailio doesnt work out from the box.

MingHon gminghon at gmail.com
Tue Jun 28 07:53:57 CEST 2011


Hi,

i fixed the audio issue for 102 to 103 vice versa.

by fixing the canreinvite in asterisk.

from uac the rtp packet will route to kamailio den forward to asterisk.

can we bypass the rtp packet going to asterisk?

and here is the update for uac 101 issue.

when 101 call to voicemail or 102/103 there is no audio.

in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.

but if 102/103 call to 101 both uac got audio.

i realize this is because 101 is the first uac registered before 102/103 and
because it did not have the received: field in ul show.

please adv.

-- 
Regards,

MingHon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110628/ed87c57a/attachment.htm>


More information about the sr-users mailing list