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 <[email protected]> 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:[email protected]: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 > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >
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