Hello
I am facing a problem as below. Please suggest for the work around.
My call flow is like this.
SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy
So when the call arrives at Kamalio+RTPProxy, i m getting below in log.
Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@
192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong
Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@
192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1
Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with
192.168.100.177:27360Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@
192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with
192.168.100.177:27361
But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179
What can be done to change the caller's RTP address to x.179 in RTPProxy ?
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Regards
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Sujit Roy
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