Is there any possibility to use external RTP rpoxy?

Asterisk Asterisk SIP
1

Is there any possibility to use external RTP rpoxy for asterisk or make it they way it will not touch packets between endpeers?

Found that there is packet2packet bridge, but no glue how to set it up.
Tried also sip options:
canreinvite=no
directrtpsetup=yes
directmedia=yes
ignoresdpversion=yes
avpf=no

but nothing helps, asterisk still touching packets and I have each packet new SEQ number, while it should be the same for send and received if it is not touch them

Any possibility to do this?

PS: Asterisk 13

2

You have conflicting options set. The canreinvite and directmedia options are equivalent.

As well conditions have to allow direct media to occur. Encryption, different codec, recording, different DTMF, NAT, and others can prevent it. You’ll need to provide console output and more details about what exactly is going on.

3

Is there any way to pass custom a=* in sdp packets so asterisk would not remove them?
Custom sip phones passing some string via a=* param, for example a=some string, but outgoing packets from asterisk does not contains them

4

No. Asterisk is a back to back user agent and the SDP is regenerated on the other leg of the call.

5

Yeah, I know, but may be ther is a possibility to store it somehow via AMI or some module “on packet received” and then inject it back “on before response send”?