we have a asterisk server, the sip proxy works fine in UDP/TCP+ NON-SRTP mode in iPhone/Android, but when we enable the UDP+SRTP, the callee can not received any voices, while the caller works fine. i capture the network log and found no RTP packets are loaded from sip server(asterisk) to callee. i don't know if it is the NAT problem in sip server?
* in addition, i use another open source client csipsample with our asterisk server, and it works fine.
* i also set the stun server as 'stun.linphone.org' or other stun server in internet, but the problem still persists.
Any suggestions or ideas would be appreciated. Thanks!