Hi everyone, I have installed the opensips 1.6.2 on our network . it has private address 192.168.1.1. We are able to register using this server from behind our network as well as from outside our network using public ip address. The problem is I am not able to make a conection between two linphone clients which uses sip proxy. when i dug in i found out that the problem is in actually the INVITE packet being transferred between two clients. >From what i know that opensips is supposed to change the contact information in the sdp packet. so it does but it changes the contact information of the incoming INVITE packet from public IP address to private IP address i.e. 192.168.1.1. & the other client ends up sending ACK packet to this address. & thus it never reaches the first client.
Message send from a client behind our netwrk only ortp-message-Message sent: (to dest=192.168.1.1:5060) SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKc3ad.cfe10033.0 Via: SIP/2.0/UDP 59.161.177.228:5060 ;received=59.161.177.228;rport=5060;branch=z9hG4bK1023899182 Record-Route: <sip:192.168.1.1;lr=on> From: <sip:tus...@private_ip>;tag=332199248 To: <sip:tus...@private_ip>;tag=1564457814 Call-ID: 637708402 CSeq: 21 INVITE Contact: <sip:[email protected]:5060> Content-Type: application/sdp User-Agent: Linphone/3.2.1 (eXosip2/3.3.0) Content-Length: 231 v=0 o=root 123456 654321 IN IP4 192.168.1.248 s=A conversation c=IN IP4 192.168.1.248 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 0 Message received by other client outside our network. ortp-message-Received message: SIP/2.0 200 OK Via: SIP/2.0/UDP 59.161.177.228:5060 ;received=59.161.177.228;rport=5060;branch=z9hG4bK1023899182 Record-Route: <sip:192.168.1.1;lr=on> From: <sip:tus...@private_ip>;tag=332199248 To: <sip:tus...@private_ip>;tag=1564457814 Call-ID: 637708402 CSeq: 21 INVITE Contact: <sip:[email protected]:5060;nat=yes> Content-Type: application/sdp User-Agent: Linphone/3.2.1 (eXosip2/3.3.0) Content-Length: 249 P-hint: onreply_route|force_rtp_proxy P-hint: Onreply-route - fixcontact v=0 o=root 123456 654321 IN IP4 192.168.1.248 s=A conversation c=IN IP4 192.168.1.1 t=0 0 m=audio 55924 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 0 a=nortpproxy:yes & hence it send the ACK message as ortp-message-Message sent: (to dest=192.168.1.1:5060) ACK sip:[email protected]:5060;nat=yes SIP/2.0 Via: SIP/2.0/UDP 59.161.177.228:5060;rport;branch=z9hG4bK2056355642 Route: <sip:192.168.1.54;lr=on> From: <sip:tus...@private_ip>;tag=332199248 To: <sip:tus...@private_ip>;tag=1564457814 Call-ID: 637708402 CSeq: 21 ACK Contact: <sip:[email protected]:5060> Proxy-Authorization: Digest username="tuser1", realm="PRIVATE_IP5", nonce="4c0f700200000037926169c961edd9ac10258ccf5fa75912", uri="sip:tus...@private_ip" , response="00dab0a4a4a14d24a2a9a69daf5136ee", algorithm=MD5 Max-Forwards: 70 User-Agent: Linphone/3.2.1 (eXosip2/3.3.0) Content-Length: 0 & therefore this ACK never reaches our server & the call gets dropped. The same scenario stands true even if both the clients are out side our network. I have written the opensips.cfg file exactly as mentioned in the book 'Building Telephony system using Opensips1.6' by flavio gnocalves. If somebody has faced this problem please guide me in the right direction. Any help will be appreciated. -- Thanks & regards yuvraj pasi
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