What steps will reproduce the problem?
1. Build for java
2. In the java application: start the stack, register, create new CallSession
and call its callAudio method
What is the expected output? What do you see instead?
SIP session is established (INVITE, 183, PRACK, 200 (PRACK), 180, 200 (INVITE),
ACK).
Doubango starts sending RTP after receiving the first 180, and stops sending
any RTP after sending the ACK.
What version of the product are you using? On what operating system?
Version 646 on vista.
Please provide any additional information below.
See the attached log.
Original issue reported on code.google.com by kallab...@gmail.com on 19 Aug 2011 at 3:31
Original issue reported on code.google.com by
kallab...@gmail.com
on 19 Aug 2011 at 3:31Attachments: