Hi Sujit,
you can use sngrep to inspect the sdp coming from the upstream sip gateways
and the answer that Kamailio creates.
This helps you to understand and debug.
Cheers
Karsten Horsmann
Sujit Roy <sujitroydhk(a)gmail.com> schrieb am Mi., 27. Nov. 2019, 07:13:
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 192.168.100.177:27360
Nov 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 ?
Thanks
--
Regards
===================
Sujit Roy
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