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

[Narayan P]

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 at 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 at 185.122.206.62;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm
To: <sip:919035438928 at 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 at 185.122.206.62;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm
To: <sip:919035438928 at 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 at 185.122.206.62>;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm
To: <sip:919035438928 at 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 at 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 at 185.122.206.62;tag=YZDJ3TTb4kDIO2.0Bvuw8nrkTTNPBwtm
To: <sip:919035438928 at 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 at 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


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