I’m using Asterisk 1.6.0 branch
Sometimes one leg hangup unexpectedly, the other leg doesn’t hangup. It happens with different phones brands, on internal and external calls.
I’ve anylized SIP and RTP traffic:
SIP INVITE is normal
Both RTP streams start normally
When the first leg hangup unexpectedly, the associated RTP stream stop, the other one keeps going
When the user hangup the other side, the second RTP stream stop and BYE are exchanged
I’m not sure this is the problem but wireshark says there are 1 or 2 RTP wrong timestamps in the long RTP stream.
Does anyone know what can i do to solve this problem?