SIP Proxy with NAT

1. Brekeke Product Name and version:
SIP Server ver 2.4.5.5

2. Java version:
Sun java 6.2

3. OS type and the version:
Ubuntu 10.04

4. UA (phone), gateway or other hardware/software involved:
Cisco ASA

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
Pattern 3

6. Your problem:
No audio for remote UA, it is a web phone using the SIP Proxy feature of SIP server. Call is correctly forwarded to internal SIP server and connects, I see RTP stream from server to Brekeke server but it stops when leaving for remote. Firewall shows correct ports and status. Need to be able to proxy for SIP and RTP and currently it gets set up but SIP controls and RTP does not work.

does the call show as "talking" in brekeke sip server/active sessions?

http://wiki.brekeke.com/wiki/Using-Brek ... a-firewall

Yes, I think I found the issue, the SIP ALG on the ASA was holding connections open, disabled the SIP and it appears to be working now.