Hi everyone,<br>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 <br>server from behind our network as well as from outside our network using public ip address. <br>
The problem is I am not able to make a conection between two linphone clients which uses sip proxy.<br>when i dug in i found out that the problem is in actually the INVITE packet being transferred between two clients.<br>
>From what i know that opensips is supposed to change the contact information in the sdp packet. so it does but <br>it changes the contact information of the incoming INVITE packet from public IP address to private IP address i.e. <br>
192.168.1.1. & the other client ends up sending ACK packet to this address. & thus it never reaches the first client.<br><br>Message send from a client behind our netwrk only<br><br>ortp-message-Message sent: (to dest=<a href="http://192.168.1.1:5060">192.168.1.1:5060</a>)<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKc3ad.cfe10033.0<br>Via: SIP/2.0/UDP 59.161.177.228:5060;received=59.161.177.228;rport=5060;branch=z9hG4bK1023899182<br>Record-Route: <sip:192.168.1.1;lr=on><br>
From: <sip:tuser1@PRIVATE_IP>;tag=332199248<br>To: <sip:tuser2@PRIVATE_IP>;tag=1564457814<br>Call-ID: 637708402<br>CSeq: 21 INVITE<br>Contact: <<a href="http://sip:tuser2@192.168.1.248:5060">sip:tuser2@192.168.1.248:5060</a>><br>
Content-Type: application/sdp<br>User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)<br>Content-Length: 231<br><br>v=0<br>o=root 123456 654321 IN IP4 192.168.1.248<br>s=A conversation<br>c=IN IP4 192.168.1.248<br>t=0 0<br>m=audio 7078 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000/1<br>a=rtpmap:8 PCMA/8000/1<br>a=rtpmap:101 telephone-event/8000<br>m=video 0 RTP/AVP 0<br><br><br>Message received by other client outside our network.<br><br>ortp-message-Received message: <br>SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 59.161.177.228:5060;received=59.161.177.228;rport=5060;branch=z9hG4bK1023899182<br>Record-Route: <sip:192.168.1.1;lr=on><br>From: <sip:tuser1@PRIVATE_IP>;tag=332199248<br>To: <sip:tuser2@PRIVATE_IP>;tag=1564457814<br>
Call-ID: 637708402<br>CSeq: 21 INVITE<br>Contact: <sip:tuser2@192.168.1.248:5060;nat=yes><br>Content-Type: application/sdp<br>User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)<br>Content-Length: 249<br>P-hint: onreply_route|force_rtp_proxy <br>
P-hint: Onreply-route - fixcontact <br><br>v=0<br>o=root 123456 654321 IN IP4 192.168.1.248<br>s=A conversation<br>c=IN IP4 192.168.1.1<br>t=0 0<br>m=audio 55924 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000/1<br>a=rtpmap:8 PCMA/8000/1<br>
a=rtpmap:101 telephone-event/8000<br>m=video 0 RTP/AVP 0<br>a=nortpproxy:yes<br><br><br>& hence it send the ACK message as<br><br>ortp-message-Message sent: (to dest=<a href="http://192.168.1.1:5060">192.168.1.1:5060</a>)<br>
ACK sip:tuser2@192.168.1.248:5060;nat=yes SIP/2.0<br>Via: SIP/2.0/UDP 59.161.177.228:5060;rport;branch=z9hG4bK2056355642<br>Route: <sip:192.168.1.54;lr=on><br>From: <sip:tuser1@PRIVATE_IP>;tag=332199248<br>To: <sip:tuser2@PRIVATE_IP>;tag=1564457814<br>
Call-ID: 637708402<br>CSeq: 21 ACK<br>Contact: <<a href="http://sip:tuser1@59.161.177.228:5060">sip:tuser1@59.161.177.228:5060</a>><br>Proxy-Authorization:
Digest username="tuser1", realm="PRIVATE_IP5",
nonce="4c0f700200000037926169c961edd9ac10258ccf5fa75912",
uri="sip:tuser2@PRIVATE_IP"<br>, response="00dab0a4a4a14d24a2a9a69daf5136ee", algorithm=MD5<br>Max-Forwards: 70<br>User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)<br>Content-Length: 0<br><br clear="all">& therefore this ACK never reaches our server & the call gets dropped.<br>
<br>
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 <br>
'Building Telephony system using Opensips1.6' by flavio gnocalves.<br>
<br>
If somebody has faced this problem please guide me in the right direction.<br>
Any help will be appreciated.<br clear="all"><br>-- <br>Thanks & regards<br>yuvraj pasi<br>