One way audio due to port change from initial SDP Invite

shagemeyer · Post by **shagemeyer** » Wed Dec 05, 2012 8:13 am

1. Brekeke Product Name and Version: SIP Server 3.0.7.0

2. Java version: 7.9

3. OS type and the version: Windows 2008 R2

4. UA (phone), gateway or other hardware/software involved:
Rauland Nurse call device (Branch Regional Controller), Bria Softphone, Cisco Wireless Handset

5. Your problem: The initial sip invite is the following:
10.165.252.43 port 5004 contained in the message body.
When the call is connected, the port changes to the port that the system connected to 10.165.252.43 to communicate data on, not the initial invite port. I am relaying off the SIP server and every other device has two way audio except for the patient stations. Why is the audio port not staying with the initial invite port of 5004. If I tell the SIP server to use the SDP port, it uses the "Rauland Gateway" server IP of 165.68.169.93 and port 5004 instead of 10.165.252.43 port 5004. I have audio out of the device.

davi · Post by **davi** » Wed Dec 05, 2012 5:49 pm

Try the settings below:

In the [Configuration]-[SDP] page,
[RTP relay] = on
[Port mapping]= sdp
[Send UA's remote address] = no