[SR-Users] Kamailio: how to route RTP media directly to endpoint

[Daniel Tryba]

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.