Hi List,<div><br></div><div>im facing an issue that my kamailio proxy did not replace the ip address in the invite and 200OK sdp body.</div><div><br></div><div>my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user</div>
<div><br></div><div>my kamailio is listening on 192.168.1.3, also define: advertised_address="175.136.223.112"; & advertised_port=5060;</div><div><br></div><div>and my asterisk is on 192.168.1.23.</div><div>
<br></div><div>sip signalling and rtp port forwarded to kamailio.</div><div><br></div><div>uacs from another nat register successfully.</div><div><br></div><div>if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112");</div>
<div><br></div><div>i will get double ip addr in c and o but kamailio ignore my ip addr. example i will get </div><div><br></div><div><div>c=IN IP4 192.168.1.3192.168.1.3</div></div><div><br></div><div>here is part of my simple script.</div>
<div><br></div><div>hope you can help.</div><div><br></div><div>thank you very much.</div><div><br></div><div>---------------cfg-------------------</div><div><br></div><div><div>route[RTPPROXY] {</div><div>#!ifdef WITH_NAT</div>
<div><span class="Apple-tab-span" style="white-space:pre"> </span>if (is_method("BYE")) {</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>unforce_rtp_proxy();</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>} else if (is_method("INVITE")){</div>
<div><span class="Apple-tab-span" style="white-space:pre"> </span>force_rtp_proxy("fcow","175.136.223.112");</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>#force_rtp_proxy("fcow","175.136.223.112");</div>
<div><span class="Apple-tab-span" style="white-space:pre"> </span>xlog("L_INFO","offer");</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>}</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>if (!has_totag()) add_rr_param(";nat=yes");</div>
<div>#!endif</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>return;</div><div>}</div></div><div><br></div><div>--------------------------------------</div><div><br></div><div>and here is the wireshark for uac INVITE and OK.</div>
<div><br></div><div>-----------INVITE-----------------</div><div><br></div><div><div>ve0</div><div>EE;p9INVITE <a href="http://sip:102@192.168.2.132:5062">sip:102@192.168.2.132:5062</a> SIP/2.0</div><div>Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes></div>
<div>Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0</div><div>Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080</div><div>Max-Forwards: 69</div><div>From: "101" <<a href="mailto:sip%3A102@aextddns.dyndns.info">sip:102@aextddns.dyndns.info</a>>;tag=as032358a3</div>
<div>To: <<a href="http://sip:102@192.168.1.3:5060">sip:102@192.168.1.3:5060</a>></div><div>Contact: <<a href="http://sip:102@192.168.1.23:5080">sip:102@192.168.1.23:5080</a>></div><div>Call-ID: <a href="mailto:416f6e09674ae9671bb7144a1cb11137@aextddns.dyndns.info">416f6e09674ae9671bb7144a1cb11137@aextddns.dyndns.info</a></div>
<div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX 1.6.2.18</div><div>Date: Tue, 05 Jul 2011 07:20:53 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</div><div>Supported: replaces, timer</div>
<div>Content-Type: application/sdp</div><div>Content-Length: 327</div><div><br></div><div>v=0</div><div>o=root 1639709788 1639709788 IN IP4 192.168.1.3</div><div>s=Asterisk PBX 1.6.2.18</div><div>c=IN IP4 192.168.1.3</div>
<div>t=0 0</div><div>m=audio 10072 RTP/AVP 0 3 8 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>
a=silenceSupp:off - - - -</div><div>a=ptime:20</div><div>a=sendrecv</div><div>a=nortpproxy:yes</div></div><div><br></div><div>-----------200OK---------------</div><div><div><div><br></div></div></div><div><div>e90</div><div>
ElE;pX4tSIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP 192.168.2.200:5062;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416</div><div>Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes></div><div>From: "101" <<a href="mailto:sip%3A101@aextddns.dyndns.info">sip:101@aextddns.dyndns.info</a>>;tag=1796959074</div>
<div>To: <<a href="mailto:sip%3A102@aextddns.dyndns.info">sip:102@aextddns.dyndns.info</a>>;tag=as2e4c0125</div><div>Call-ID: <a href="mailto:1985782590@192.168.2.200">1985782590@192.168.2.200</a></div><div>CSeq: 21 INVITE</div>
<div>Server: Asterisk PBX 1.6.2.18</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</div><div>Supported: replaces, timer</div><div>Contact: <<a href="http://sip:102@192.168.1.23:5080">sip:102@192.168.1.23:5080</a>></div>
<div>Content-Type: application/sdp</div><div>Content-Length: 286</div><div><br></div><div>v=0</div><div>o=root 403900934 403900934 IN IP4 192.168.1.23</div><div>s=Asterisk PBX 1.6.2.18</div><div>c=IN IP4 192.168.1.23</div>
<div>t=0 0</div><div>m=audio 14420 RTP/AVP 0 8 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div>
<div>a=ptime:20</div><div>a=sendrecv</div></div><div><br></div><div>------------------------------------</div><div><br></div><div>My kamailio log.</div><div><br></div><div>-----------LOG------------------</div><div><br></div>
<div><div>DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid</div><div>DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3</div><div>INFO: <script>: offer</div></div><div>
<br></div><div>-------------------------------------</div><div><br></div><div>double force_rtp_proxy</div><div><br></div><div>--------kamailio -> asterisk [INVITE]---------</div><div><br></div><div><div>Pyi-}E7V@:#pINVITE <a href="mailto:sip%3A102@aextddns.dyndns.info">sip:102@aextddns.dyndns.info</a> SIP/2.0</div>
<div>Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes></div><div>Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0</div><div>Via: SIP/2.0/UDP 192.168.2.200:5062;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648</div>
<div>From: "101" <<a href="mailto:sip%3A101@aextddns.dyndns.info">sip:101@aextddns.dyndns.info</a>>;tag=640933430</div><div>To: <<a href="mailto:sip%3A102@aextddns.dyndns.info">sip:102@aextddns.dyndns.info</a>></div>
<div>Call-ID: <a href="mailto:1909950509@192.168.2.200">1909950509@192.168.2.200</a></div><div>CSeq: 21 INVITE</div><div>Contact: <<a href="http://sip:101@175.138.21.31:2788">sip:101@175.138.21.31:2788</a>></div><div>
Content-Type: application/sdp</div><div>Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE</div><div>Max-Forwards: 69</div><div>User-Agent: T20 9.41.0.80</div>
<div>Allow-Events: talk,hold,conference,refer,check-sync</div><div>Content-Length: 334</div><div><br></div><div>v=0</div><div>o=20073 20073 IN IP4 192.168.1.3192.168.1.3</div><div>s=SDP data</div><div>c=IN IP4 192.168.1.3192.168.1.3</div>
<div>t=0 0</div><div>m=audio 1006410064 RTP/AVP 0 8 18 9 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:18 G729/8000</div><div>a=rtpmap:9 G722/8000</div><div>a=fmtp:101 0-15</div><div>
a=rtpmap:101 telephone-event/8000</div><div>a=sendrecv</div><div>a=nortpproxy:yes</div><div>a=nortpproxy:yes</div></div><div><br></div><div>-----------LOG------------------</div><div><br></div><div><div>DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid</div>
<div>DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3</div><div>DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid</div><div>DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3</div>
<div>INFO: <script>: offer</div></div><div><br></div><div>-----------LOG------------------</div><div><br></div><div><br></div><div>-- </div><div>Regards,<br><br>MingHon<br>
</div>