[SR-Users] How to convert ipv6 to ipv4 address inside message body

Hi, I am using kamailio as outbound proxy integrated with rtpproxy.I have done the configuration of ipv6 to ipv4 network. But when 183 session progress came ,kamailio is sending its ipv6 address to pjsip client inside sdp body.So that it is not accepting and 488 error message is giving. How to convert ipv6 to ipv4 inside kamailio so that pjsip can accept. Below is the sip trace from client. 20:49:20.152 pjsua_app.c .....Call 0 state changed to EARLY (183 Session progress) 20:49:20.152 pjsua_media.c .....Call 0: updating media.. 20:49:20.152 pjsua_media.c ......pjmedia_stream_info_from_sdp() failed for call_id 0 media 0: Unsupported address family (PJ_EAFNOTSUP) 20:49:20.152 pjsua_media.c ......Error updating media call00:0: Unsupported address family (PJ_EAFNOTSUP) 20:49:20.152 pjsua_call.c .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048] 20:49:20.152 pjsua_core.c ........TX 418 bytes Request msg CANCEL/cseq=12308 (tdta0x7f0b28008400) to UDP 2001:db8:bad:a55::2:5070: CANCEL sip:919035438928@185.122.206.62 SIP/2.0 Via: SIP/2.0/UDP [2001:DB8:BAD:A55:0:0:0:2]:5077;rport;branch=z9hG4bKPj8pbyM6u7jh2tBncOqhPSbPMA8mK4RaIz Max-Forwards: 70 From: sip:OTT919620649614@185.122.206.62;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm To: <sip:919035438928@185.122.206.62> Call-ID: 7otfmBr7j6wvfhjlApd0xcHOPp6wsII6 CSeq: 12308 CANCEL Route: <sip:[2001:db8:bad:a55::2]:5070;lr> Content-Length: 0 --end msg-- 20:49:20.154 pjsua_core.c .RX 457 bytes Response msg 200/CANCEL/cseq=12308 (rdata0x7f0b28001d08) from UDP 2001:db8:bad:a55::2:5070: SIP/2.0 200 canceling Via: SIP/2.0/UDP [2001:DB8:BAD:A55:0:0:0:2]:5077;rport=5077;branch=z9hG4bKPj8pbyM6u7jh2tBncOqhPSbPMA8mK4RaIz;received=2001:DB8:BAD:A55:0:0:0:2 From: sip:OTT919620649614@185.122.206.62;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm To: <sip:919035438928@185.122.206.62>;tag=6374de6d95f421336d5241fc0a072275-2b9b Call-ID: 7otfmBr7j6wvfhjlApd0xcHOPp6wsII6 CSeq: 12308 CANCEL Server: kamailio (5.0.0-dev5 (x86_64/linux)) Content-Length: 0 --end msg-- 20:49:20.310 pjsua_core.c .RX 380 bytes Response msg 487/INVITE/cseq=12308 (rdata0x7f0b28001d08) from UDP 2001:db8:bad:a55::2:5070: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [2001:DB8:BAD:A55:0:0:0:2]:5077;rport=5077;branch=z9hG4bKPj8pbyM6u7jh2tBncOqhPSbPMA8mK4RaIz;received=2001:DB8:BAD:A55:0:0:0:2 From: <sip:OTT919620649614@185.122.206.62>;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm To: <sip:919035438928@185.122.206.62> CSeq: 12308 INVITE Call-ID: 7otfmBr7j6wvfhjlApd0xcHOPp6wsII6 Content-Length: 0 --end msg-- 20:49:20.310 pjsua_core.c ..TX 412 bytes Request msg ACK/cseq=12308 (tdta0x7f0b2800cf40) to UDP 2001:db8:bad:a55::2:5070: ACK sip:919035438928@185.122.206.62 SIP/2.0 Via: SIP/2.0/UDP [2001:DB8:BAD:A55:0:0:0:2]:5077;rport;branch=z9hG4bKPj8pbyM6u7jh2tBncOqhPSbPMA8mK4RaIz Max-Forwards: 70 From: sip:OTT919620649614@185.122.206.62;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm To: <sip:919035438928@185.122.206.62> Call-ID: 7otfmBr7j6wvfhjlApd0xcHOPp6wsII6 CSeq: 12308 ACK Route: <sip:[2001:db8:bad:a55::2]:5070;lr> Content-Length: 0 --end msg-- 20:49:20.311 pjsua_app.c .....Call 0 is DISCONNECTED [reason=488 (Not Acceptable Here)] 20:49:20.311 pjsua_app_comm ..... [DISCONNCTD] To: <sip:919035438928@185.122.206.62>;tag=GB-1498835320600 Call time: 00h:00m:00s, 1st res in 361 ms, conn in 0ms #0 audio deactivated 20:49:20.311 pjsua_media.c .....Call 0: deinitializing media.. 20:49:20.311 pjsua_media.c .......Media stream call00:0 is destroyed 20:49:21.311 pjsua_aud.c !Closing sound device after idle for 1 second(s) 20:49:21.311 pjsua_app.c .Turning sound device OFF 20:49:21.311 pjsua_aud.c .Closing default sound playback device and default sound capture device Any suggestion will be highly appreciated. Thanks, Narayan