I’ve got a test server with two IPs (one IP is virtual and moves to a backup server if the first goes down). I’m trying out PJSIP and specified the virtual IP for the Bind address of all transports I’m using.
All SIP packets are now sent of the virtual IP. But the RTP UDP data is being sent out of the wrong IP. It seems that Asterisk is ignoring the Bind address when sending out the audio.
Has anyone come across this?
I’ve tried setting external_media_address but note that this is only used in cases using NAT, which I’m not.
Many thanks Dan