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Hello,<br>
<br>
I don't understand what you want to do, however, if you look to
change content of the sdp, see the nathelper, rtpproxy, mangler,
sdpops, textops, textopsx and siputils modules -- these should give
you plenty of options to play with it to update the sdp.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 20/11/14 18:21, Javier Ricke wrote:<br>
</div>
<blockquote
cite="mid:CAP7s5+zsKpffVUezF5NB=LQKh1C3bH8WufCoXuaf+Ubhx_9Tmw@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>------phona A-------kamailio---------asterisk-----</div>
<div>OPTION 1: configure asterisk or kamailio</div>
<div>i used asterisk, and install kamailio for traffic RTP can
be send between end points that behind NAT router, and do not
have to go through RTP proxy,, plz help!</div>
<div>i think to the moment install kamailio, headrs'sdp fix IP
private, but no!, how can fix it plz!!?</div>
<div>regards!</div>
<div><br>
</div>
<div>or </div>
<div>OPTION 2: edit sip/sdp</div>
<div>mi sip/sdp is</div>
<div>[code]</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://152.74.21.12:5060">152.74.21.12:5060</a>
---></div>
<div>ACK <a moz-do-not-send="true"
href="http://sip:1001@152.74.21.12:6112">sip:1001@152.74.21.12:6112</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP 152.74.21.12;branch=z9hG4bKcydzigwkX</div>
<div>Via: SIP/2.0/UDP
190.164.204.227:41553;branch=z9hG4bK-d8754z-84f73c7e042445de-1---d8754z-;rport=41553</div>
<div>Max-Forwards: 16</div>
<div>Contact:
<a class="moz-txt-link-rfc2396E" href="sip:JavierTren@190.164.204.227:41553;transport=UDP"><sip:JavierTren@190.164.204.227:41553;transport=UDP></a></div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A1001@152.74.21.12">sip:1001@152.74.21.12</a>;transport=UDP>;tag=as6e487bf1</div>
<div>From: <<a moz-do-not-send="true"
href="mailto:sip%3AJavierTren@152.74.21.12">sip:JavierTren@152.74.21.12</a>;transport=UDP>;tag=2b8fa52c</div>
<div>Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.</div>
<div>CSeq: 2 ACK</div>
<div>Proxy-Authorization: Digest
username="JavierTren",realm="152.74.21.12",nonce="VGFnblRhZkIpJRpScSaEi795VKe4uof0",uri="<a
moz-do-not-send="true" href="mailto:sip%3A1001@152.74.21.12">sip:1001@152.74.21.12</a>;transport=UDP",response="116bf459c22231d0a770534d674b768d",algorithm=MD5</div>
<div>User-Agent: Zoiper r27147</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><-------------></div>
<div>--- (12 headers 0 lines) ---</div>
<div>set_destination: Parsing
<a class="moz-txt-link-rfc2396E" href="sip:152.74.21.12;lr=on;ftag=2b8fa52c"><sip:152.74.21.12;lr=on;ftag=2b8fa52c></a> for address/port
to send to</div>
<div>set_destination: set destination to <a
moz-do-not-send="true" href="http://152.74.21.12:5060">152.74.21.12:5060</a></div>
<div>Audio is at 12064</div>
<div>Adding codec 100002 (gsm) to SDP</div>
<div>Adding codec 100003 (ulaw) to SDP</div>
<div>Adding non-codec 0x1 (telephone-event) to SDP</div>
<div>Reliably Transmitting (NAT) to <a moz-do-not-send="true"
href="http://152.74.21.12:5060">152.74.21.12:5060</a>:</div>
<div>INVITE <a class="moz-txt-link-freetext" href="sip:JavierTren@190.164.204.227:41553;transport=UDP">sip:JavierTren@190.164.204.227:41553;transport=UDP</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
152.74.21.12:6112;branch=z9hG4bK3992226d;rport</div>
<div>Route: <a class="moz-txt-link-rfc2396E" href="sip:152.74.21.12;lr=on;ftag=2b8fa52c"><sip:152.74.21.12;lr=on;ftag=2b8fa52c></a></div>
<div>Max-Forwards: 70</div>
<div>From: <<a moz-do-not-send="true"
href="mailto:sip%3A1001@152.74.21.12">sip:1001@152.74.21.12</a>;transport=UDP>;tag=as6e487bf1</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3AJavierTren@152.74.21.12">sip:JavierTren@152.74.21.12</a>;transport=UDP>;tag=2b8fa52c</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:1001@152.74.21.12:6112">sip:1001@152.74.21.12:6112</a>></div>
<div>Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.</div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Asterisk PBX 11.13.0</div>
<div>Session-Expires: 1800;refresher=uac</div>
<div>Min-SE: 90</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE</div>
<div>Supported: replaces, timer</div>
<div>X-asterisk-Info: SIP re-invite (External RTP bridge)</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 256</div>
<div><br>
</div>
<div>v=0</div>
<div>o=root 1842142539 1842142540 IN IP4 192.168.1.8</div>
<div>s=Asterisk PBX 11.13.0</div>
<div>c=IN IP4 192.168.1.8</div>
<div>t=0 0</div>
<div>m=audio 8000 RTP/AVP 3 0 101</div>
<div>a=rtpmap:3 GSM/8000</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>a=ptime:20</div>
<div><br>
</div>
<div>[/code]</div>
<div><br>
</div>
<div>and the field o and c i need to IP public y no private,,</div>
<div>plz any?</div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a></pre>
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