Hello,
I’m discovering * and i have a problem i am unable to deal with…
There are 2 sipphones from different manufacturer who use RTP with a packet time equal to 20ms.
When a call came with no packet time the server set it to 10ms.
I got a one way audio as the server as the packet time isn’t correctly set…
Asterisk suppress the packet time when it send the invite to the server.
call establishement
sipphone <-----------------> asterisk <---------------------> SIP server
INVITE (ptime:20ms)--------> invite(no ptime) ---------> invite(no ptime)
200 OK (no ptime) <---------- ( * ) <-------------------------200 OK (ptime:10ms)
I didn’t found how to set a ptime value in asterisk.
I need help
(sorry for my english)