On Wed, Aug 09, 2017 at 04:48:02PM +0300, wsotest.512 wrote:
> UserA ---sip--> Kamailio --> Asterisk --> UserB
> \-rtp--> --> --> UserB
> Is it possible at all? Maybe someone already did it .
It should work, but Asterisk is broken in this respect and may break
The root cause is that Asterisk is initially handling RTP and later
tries to reINVITE both legs with the ip of the rtpengine/userb for
media. If the ids of codecs/dtmf don't match in the m=audio SDP line RTP
will break. There is no way to get Asterisk not to handle initial RTP
and no way to not have Asterisk reINVITE if the ids differ.
Kamailio (SER) - Users Mailing List