SIP problem wit ptime


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)

What version of Asterisk are you running?

i am running * 1.2.10 with a linux kernel 2.6

Well for my researches, asterisk seems to be hardcoded with a ptime = 20ms.

anyone to confirm?

I have found a patch

I had installed asterisk from the tar, how to apply this patch?