9 Aug
2017
9 Aug
'17
10:29 a.m.
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
codecs/dtmf: https://issues.asterisk.org/jira/browse/ASTERISK-25166
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.