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Raviprakash,<br>
<br>
Your SDP messages are using private IPs for the RTP stream which
carries the voice traffic. This causing the media stream to be sent to
the wrong IP address, thus no audio.<br>
<br>
Have you read through and tried to follow the directions in these
documents?<br>
<br>
<ul>
  <li class="level1">
    <div class="li"> <strong><em><span style="color: teal;">OpenSER
&amp; NAT</span></em></strong></div>
    <ul>
      <li class="level1">
        <div class="li"> <a
 href="http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy"
 class="wikilink1" title="nat:remote-rtpproxy">Run RTPproxy on a remote
host</a></div>
      </li>
      <li class="level1">
        <div class="li">&nbsp;&nbsp;&nbsp; <a
 href="http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy"
 class="wikilink1" title="nat:remote-rtpproxy">http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy</a><a
 href="http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy"
 class="wikilink1" title="nat:remote-rtpproxy"></a></div>
      </li>
      <li class="level1">
        <div class="li"> <a
 href="http://voip-info.org/wiki/view/OpenSER+And+RTPProxy"
 class="urlextern"
 title="http://voip-info.org/wiki/view/OpenSER+And+RTPProxy"
 rel="nofollow">OpenSER and RTPProxy</a></div>
      </li>
      <li class="level1">
        <div class="li">&nbsp;&nbsp;&nbsp; <a
 href="http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy"
 class="wikilink1" title="nat:remote-rtpproxy"><u><a
 href="http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy"
 class="wikilink1" title="nat:remote-rtpproxy">http://voip-info.org/wiki/view/OpenSER+And+RTPProxy</a></u></a></div>
      </li>
      <li class="level1">
        <div class="li"> <a
 href="http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy"
 class="urlextern"
 title="http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy"
 rel="nofollow"> OpenSER and Mediaproxy <br>
        </a></div>
      </li>
      <li class="level1">
        <div class="li">&nbsp;&nbsp;&nbsp; <a
 href="http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy"
 class="urlextern"
 title="http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy"
 rel="nofollow">http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy</a></div>
      </li>
    </ul>
  </li>
</ul>
A lot of good documentation has been written up on the OpenSER
application which can be found at this website.<br>
<br>
&nbsp;&nbsp;&nbsp; <a class="moz-txt-link-freetext" href="http://openser.org/dokuwiki/doku.php">http://openser.org/dokuwiki/doku.php</a><br>
<br>
raviprakash sunkara wrote:
<blockquote
 cite="midf0f0f9340702210108gc9a1c03hd0d0861d6329e7e9@mail.gmail.com"
 type="cite">Hello Users,
  <br>
  <br>
I posted&nbsp; so many mail to users but no one reply my issue please&nbsp; help&nbsp;
me
  <br>
  <br>
openSER proxy is mysipdomain.com , and its private_ ip is 192.168.2.60
and
  <br>
SIP server and Proxy is also in Behind
  <br>
UAC's are Behind the NATs
  <br>
  <br>
--------------------- INVITE ---------------
  <br>
U 59.144.88.7:5060 -&gt; 192.168.2.60:5060
  <br>
INVITE <a class="moz-txt-link-freetext" href="sip:9002@mysipdomain.com;user=phone">sip:9002@mysipdomain.com;user=phone</a> SIP/2.0.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKe2a540a8170eb12a.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 1 INVITE.
  <br>
Contact: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@192.168.1.2:5060;user=phone;transport=udp">&lt;sip:8002@192.168.1.2:5060;user=phone;transport=udp&gt;</a>.
  <br>
User-Agent: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Expires: 300.
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Supported: 100rel,replaces.
  <br>
Content-Length: 245.
  <br>
Content-Type: application/sdp.
  <br>
v=0.
  <br>
  <font color="#ff0000"><big><b>o=8002 14279 14279 IN IP4 <u>192.168.1.2.</u></b></big></font><u>
  </u><br>
s=ATA186 Call.
  <br>
c=IN IP4 192.168.1.2.
  <br>
t=0 0.
  <br>
m=audio 16386 RTP/AVP 0 4 8 101.
  <br>
a=rtpmap:0 PCMU/8000/1.
  <br>
a=rtpmap:4 G723/8000/1.
  <br>
a=rtpmap:8 PCMA/8000/1.
  <br>
a=rtpmap:101 telephone-event/8000.
  <br>
a=fmtp:101 0-15.
  <br>
  <br>
#
  <br>
U 192.168.2.60:5060 -&gt; 61.17.248.68:3186
  <br>
INVITE <a class="moz-txt-link-freetext" href="sip:9002@192.168.2.7:5060;user=phone;transport=udp">sip:9002@192.168.2.7:5060;user=phone;transport=udp</a> SIP/2.0.
  <br>
Max-Forwards: 10.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Contact: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@59.144.88.7:5060;user=phone;transport=udp">&lt;sip:8002@59.144.88.7:5060;user=phone;transport=udp&gt;</a>.
  <br>
User-Agent: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Expires: 300.
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Supported: 100rel,replaces.
  <br>
Content-Length: 265.
  <br>
Content-Type: application/sdp.
  <br>
v=0.
  <br>
o=8002 14329 14329 IN IP4 192.168.1.2.
  <br>
s=ATA186 Call.
  <br>
c=IN IP4 192.168.1.2.
  <br>
t=0 0.
  <br>
m=audio 16386 RTP/AVP 0 4 8 101.
  <br>
a=rtpmap:0 PCMU/8000/1.
  <br>
a=rtpmap:4 G723/8000/1.
  <br>
a=rtpmap:8 PCMA/8000/1.
  <br>
a=rtpmap:101 telephone-event/8000.
  <br>
a=fmtp:101 0-15.
  <br>
a=direction:active.
  <br>
  <br>
------------------- RINGING------------
  <br>
U 61.17.248.68:3186 -&gt; 192.168.2.60:5060
  <br>
SIP/2.0 180 Ringing.
  <br>
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>;tag=1332822912.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Require: 100rel.
  <br>
RSeq: 1.
  <br>
Contact: 9002
<a class="moz-txt-link-rfc2396E" href="sip:9002@192.168.2.7:5060;user=phone;transport=udp">&lt;sip:9002@192.168.2.7:5060;user=phone;transport=udp&gt;</a>.
  <br>
Server: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Content-Length: 0.
  <br>
.
  <br>
  <br>
#
  <br>
U 192.168.2.60:5060 -&gt; 59.144.88.7:5060
  <br>
SIP/2.0 180 Ringing.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>;tag=1332822912.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Require: 100rel.
  <br>
RSeq: 1.
  <br>
Contact: 9002
<a class="moz-txt-link-rfc2396E" href="sip:9002@61.17.248.68:3186;user=phone;transport=udp">&lt;sip:9002@61.17.248.68:3186;user=phone;transport=udp&gt;</a>.
  <br>
Server: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Content-Length: 0.
  <br>
  <br>
U 61.17.248.68:3186 -&gt; 192.168.2.60:5060
  <br>
SIP/2.0 200 OK.
  <br>
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>;tag=1332822912.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Contact: 9002
<a class="moz-txt-link-rfc2396E" href="sip:9002@192.168.2.7:5060;user=phone;transport=udp">&lt;sip:9002@192.168.2.7:5060;user=phone;transport=udp&gt;</a>.
  <br>
Server: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Supported: replaces.
  <br>
Content-Length: 193.
  <br>
Content-Type: application/sdp.
  <br>
.
  <br>
v=0.
  <br>
  <font color="#ff0000"><big><b>o=9002 27865 27865 IN IP4 <u>192.168.2.7</u></b></big></font><u>.
  </u><br>
s=ATA186 Call.
  <br>
c=IN IP4 192.168.2.7.
  <br>
t=0 0.
  <br>
m=audio 16386 RTP/AVP 0 101.
  <br>
a=rtpmap:0 PCMU/8000/1.
  <br>
a=rtpmap:101 telephone-event/8000.
  <br>
a=fmtp:101 0-15.
  <br>
  <br>
#
  <br>
U 192.168.2.60:5060 -&gt; 59.144.88.7:5060
  <br>
SIP/2.0 200 OK.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>;tag=1332822912.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Contact: 9002
<a class="moz-txt-link-rfc2396E" href="sip:9002@61.17.248.68:3186;user=phone;transport=udp">&lt;sip:9002@61.17.248.68:3186;user=phone;transport=udp&gt;</a>.
  <br>
Server: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Supported: replaces.
  <br>
Content-Length: 193.
  <br>
Content-Type: application/sdp.
  <br>
.
  <br>
v=0.
  <br>
o=9002 27865 27865 IN IP4 192.168.2.7.
  <br>
s=ATA186 Call.
  <br>
c=IN IP4 192.168.2.7.
  <br>
t=0 0.
  <br>
m=audio 16386 RTP/AVP 0 101.
  <br>
a=rtpmap:0 PCMU/8000/1.
  <br>
a=rtpmap:101 telephone-event/8000.
  <br>
a=fmtp:101 0-15.
  <br>
  <br>
#
  <br>
U 61.17.248.68:3186 -&gt; 192.168.2.60:5060
  <br>
SIP/2.0 200 OK.
  <br>
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>;tag=1332822912.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Contact: 9002
<a class="moz-txt-link-rfc2396E" href="sip:9002@192.168.2.7:5060;user=phone;transport=udp">&lt;sip:9002@192.168.2.7:5060;user=phone;transport=udp&gt;</a>.
  <br>
Server: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Supported: replaces.
  <br>
Content-Length: 193.
  <br>
Content-Type: application/sdp.
  <br>
.
  <br>
v=0.
  <br>
o=9002 27865 27865 IN IP4 192.168.2.7.
  <br>
s=ATA186 Call.
  <br>
c=IN IP4 192.168.2.7.
  <br>
t=0 0.
  <br>
m=audio 16386 RTP/AVP 0 101.
  <br>
a=rtpmap:0 PCMU/8000/1.
  <br>
a=rtpmap:101 telephone-event/8000.
  <br>
a=fmtp:101 0-15.
  <br>
  <br>
#
  <br>
U 192.168.2.60:5060 -&gt; 59.144.88.7:5060
  <br>
SIP/2.0 200 OK.
  <br>
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
  <br>
;branch=z9hG4bKbd027751c869b9ff.
  <br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.2.60;lr=on;ftag=4240982537">&lt;sip:192.168.2.60;lr=on;ftag=4240982537&gt;</a>.
  <br>
From: Indian-2
<a class="moz-txt-link-rfc2396E" href="sip:8002@mysipdomain.com;user=phone">&lt;sip:8002@mysipdomain.com;user=phone&gt;</a>;tag=4240982537.
  <br>
To: <a class="moz-txt-link-rfc2396E" href="sip:9002@mysipdomain.com;user=phone">&lt;sip:9002@mysipdomain.com;user=phone&gt;</a>;tag=1332822912.
  <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1685867393@192.168.1.2">1685867393@192.168.1.2</a>.
  <br>
CSeq: 2 INVITE.
  <br>
Contact: 9002
<a class="moz-txt-link-rfc2396E" href="sip:9002@61.17.248.68:3186;user=phone;transport=udp">&lt;sip:9002@61.17.248.68:3186;user=phone;transport=udp&gt;</a>.
  <br>
Server: Cisco ATA 188&nbsp; v3.2.1 atasip (050616A).
  <br>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER,
PRACK,
  <br>
UPDATE.
  <br>
Supported: replaces.
  <br>
Content-Length: 193.
  <br>
Content-Type: application/sdp.
  <br>
v=0.
  <br>
o=9002 27865 27865 IN IP4 192.168.2.7.
  <br>
s=ATA186 Call.
  <br>
c=IN IP4 192.168.2.7.
  <br>
t=0 0.
  <br>
m=audio 16386 RTP/AVP 0 101.
  <br>
a=rtpmap:0 PCMU/8000/1.
  <br>
a=rtpmap:101 telephone-event/8000.
  <br>
a=fmtp:101 0-15.
  <br>
  <br>
  <pre wrap="">
<hr size="4" width="90%">
_______________________________________________
Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Users@openser.org">Users@openser.org</a>
<a class="moz-txt-link-freetext" href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users</a>
  </pre>
</blockquote>
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