Hi , But the ACK has as Route hdr the IP of your server, so the caller part should send the ACK to your proxy (and not to the RURI indication) - your opensips is the one that will do routing based on RURI and send to callee.
Where exactly is the ACK lost ? between caller and proxy or between proxy and callee ? Regards, Bogdan Yuvraj Pasi wrote: > 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 > <http://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 > <http://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 > <http://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 > <http://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 > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
