Hello everybody, I installed and configured opensips with a residential configuration setting NAT Traversal and Multidomain options with the opensips configuration tool. When I try to make a call with two users, the ACK method of the caller party never reaches the called party.
This are the messages received in the called party: INVITE sip:ralvarez@domain;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.151.199.52:45084 ;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:2427001@201.255.1.30:45084;transport=udp> To: <sip:ralvarez@domain> From: <sip:2427001@domain>;tag=2627e587 Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="2427001",realm=:"domain",nonce="4ffbbd42000000478de4c68512245fee27f7dadd4549c048",uri="sip:ralvarez@domain ;transport=udp",response="8fe20af4d2a381cc6a4ebd3ac1162f69",algorithm=MD5 Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 410 v=0 o=- 12986371568156250 1 IN IP4 10.151.199.52 s=CounterPath X-Lite 4.1 c=IN IP4 10.151.166.52 t=0 0 a=ice-ufrag:6079ef a=ice-pwd:8febe129544f5ec86474270bef1a9214 m=audio 53146 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 10.151.199.52 53146 typ host a=candidate:1 2 UDP 659134 10.151.199.52 53147 typ host SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 10.151.199.52:45084 ;received=201.255.1.30;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport=45084 To: <sip:ralvarez@domain> From: <sip:2427001@domain>;tag=2627e587 Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ. CSeq: 2 INVITE Server: OpenSIPS (1.8.0-notls (i386/linux)) Content-Length: 0 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.151.199.52:45084 ;received=201.255.1.30;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport=45084 Record-Route: <sip:192.100.1.220;lr;did=af5.ad45d1c6;nat=yes> *Contact: <sip:ralvarez@192.100.1.1:30612;rinstance=e0b11e2eb514ba1c>* To: <sip:ralvarez@domain>;tag=e637fa0f From: <sip:2427001@domain>;tag=2627e587 Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ. CSeq: 2 INVITE User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.151.199.52:45084 ;received=201.255.1.30;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport=45084 Record-Route: <sip:192.100.1.220;lr;did=af5.ad45d1c6;nat=yes> *Contact: <sip:ralvarez@192.100.1.1:30612;rinstance=e0b11e2eb514ba1c>* To: <sip:ralvarez@domain>;tag=e637fa0f From: <sip:2427001@domain>;tag=2627e587 Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp .... More OK responses ( retries ) BYE sip:2427001@201.255.1.30:45084;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.100.1.220;branch=z9hG4bKe265.bd09554.0 Via: SIP/2.0/UDP 192.100.1.200:30612 ;received=192.168.1.200;branch=z9hG4bK-d87543-4677b2285a79346c-1--d87543-;rport=30612 Max-Forwards: 69 Contact: <sip:ralvarez@192.100.1.200:30612;rinstance=e0b11e2eb514ba1c> To: <sip:2427001@domain>;tag=2627e587 From: <sip:ralvarez@domain>;tag=e637fa0f Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ. CSeq: 2 BYE User-Agent: X-Lite release 1002tx stamp 29712 Reason: SIP;description="ACK not received" Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.100.1.220;branch=z9hG4bKe265.bd09554.0;received=24.94.9.135 Via: SIP/2.0/UDP 192.100.1.200:30612 ;received=192.100.1.200;branch=z9hG4bK-d87543-4677b2285a79346c-1--d87543-;rport=30612 Contact: <sip:2427001@201.255.1.30:45084;transport=udp> To: <sip:2427001@domain>;tag=2627e587 From: <sip:ralvarez@domain>;tag=e637fa0f Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ. CSeq: 2 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 ----------------------------------------------- I need to fix the contact header value because the IP address is behind the NAT and is not reachable from the caller party location. I don't know how to do this, reading some answers in the users mail list i found that setting the global parameter "advertised_address" in the configuration file it will solved it but I'm not sure, can anybody tell me if this is correct? Thanks for all.
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