I have a strange situation here. While call recording using Asterisk(1.2.13), the In and Out file size difference increases progressively and at the end of recording, there is greater difference in the files size. Out is smaller than In file. Therefore when i mix these two files, the mix file does not mix properly, there is lag or lead in the mixed file. I am using a MS RTC API’s SIP soft phone .
But when i use third party SIP phone like SJ phone, the problem disappears.
I dont know where this problem lies. Kindly help to solve this issue.