[Spce-user] RTP problem when peering using private IP address

[Gang Du]

Hi, I'm new to the SPCE. I'm testing mr9.0 build 1 version and got problem with RTP when peering using private IP address. The topology is as below and the problem is in Scenario 4.

Debug from A shows in scenario 4, sipwise instructs party A to send RTP stream to 172.16.1.1 when A puts the call on hold and also after A resumes the call which is causing the no audio issue because A is unable to reach the private IP address 172.16.1.1.

Can someone point me to the right direction what settings I need to change to make it work please? Thanks in advance.

         Party A
               |
               |sip trunking
               |
               |public IP: 220.X.X.X
         Firewall (1:1 NAT 220.X.X.X <-> 192.168.1.1)
               |(IP: 192.168.1.254)
               |
               |
               |rtp_ext (IP:192.168.1.1, advertised-ip: 220.X.X.X)
         SPCE
               |rtp_pri (IP: 172.16.1.1 extra_socket: 172.16.1.1: 5060)
               |
               |Peering - RTP interface: pri; Force outbound call via socket: pri
               |
               |(IP: 172.16.1.2)
         Carrier
               |
               |
               |
               |
         Party B

Scenario 1: A calls B, 2-way audio; A puts B on hold, B hears moh; A resumes the call, 2-way audio.
Scenario 2: A calls B, 2-way audio; B puts A on hold, A hears moh; B resumes the call, 2-way audio.
Scenario 3: B calls A, 2-way audio; B puts A on hold, A hears moh; B resumes the call, 2-way audio.
Scenario 4: B calls A, 2-way audio; A puts B on hold, B hears silence; A resumes the call, no audio both ways.


Kind Regards,

Gang
M: +61406177678<tel:+61406177678>

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