<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:16px"><div id="yui_3_16_0_1_1431932745537_77224">Hi</div><div id="yui_3_16_0_1_1431932745537_77224"><br></div><div id="yui_3_16_0_1_1431932745537_77224">Did you use msg_apply_changes() before relaying the INVITE ?</div><div id="yui_3_16_0_1_1431932745537_77224" dir="ltr"><a href="http://kamailio.org/docs/modules/4.1.x/modules/textopsx.html#textopsx.f.msg_apply_changes" class="" style="" id="yui_3_16_0_1_1431932745537_77284">http://kamailio.org/docs/modules/4.1.x/modules/textopsx.html#textopsx.f.msg_apply_changes</a><br></div><div id="yui_3_16_0_1_1431932745537_77224" dir="ltr"><br></div><div id="yui_3_16_0_1_1431932745537_77224" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1431932745537_77224" dir="ltr">Dragos</div><div id="yui_3_16_0_1_1431932745537_77277"><br></div>  <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_1_1431932745537_77212"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_1_1431932745537_77211"> <div dir="ltr" id="yui_3_16_0_1_1431932745537_77219"> <hr size="1" id="yui_3_16_0_1_1431932745537_77221">  <font size="2" face="Arial" id="yui_3_16_0_1_1431932745537_77315"> <b><span style="font-weight:bold;">From:</span></b> José Seabra <joseseabra4@gmail.com><br> <b><span style="font-weight: bold;">To:</span></b> Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org> <br> <b><span style="font-weight: bold;">Sent:</span></b> Monday, May 18, 2015 12:31 PM<br> <b><span style="font-weight: bold;">Subject:</span></b> [SR-Users] Function sdp_remove_codecs_by_id seems to be not working<br> </font> </div> <div class="y_msg_container" id="yui_3_16_0_1_1431932745537_77210"><br><div id="yiv9103466334"><div dir="ltr" id="yui_3_16_0_1_1431932745537_77209">Hello,<div id="yui_3_16_0_1_1431932745537_77217"><br></div><div id="yui_3_16_0_1_1431932745537_77228">I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove some codecs in INVITE request before send  it to freeswitch, but the function doesn't remove the codec, and it doesn't give any error message.</div><div><br></div><div id="yui_3_16_0_1_1431932745537_77317">I'm using this function in request route.</div><div id="yui_3_16_0_1_1431932745537_77236"><br></div><div id="yui_3_16_0_1_1431932745537_77234"><br></div><div id="yui_3_16_0_1_1431932745537_77215">Kamailio version is 4.2.2.</div><div id="yui_3_16_0_1_1431932745537_77208"><br clear="all"><div id="yui_3_16_0_1_1431932745537_77319"><br></div><div id="yui_3_16_0_1_1431932745537_77213">INVITE that kamailio receives from phone:</div><div id="yui_3_16_0_1_1431932745537_77207"><br></div><div id="yui_3_16_0_1_1431932745537_77322"><div id="yui_3_16_0_1_1431932745537_77321">INVITE <a rel="nofollow" ymailto="mailto:sip%3A401@teste.itcenter.com.pt" target="_blank" href="mailto:sip%3A401@teste.itcenter.com.pt">sip:401@teste.d</a><a rel="nofollow" target="_blank" href="http://emo.pt/">emo.pt</a>;user=phone SIP/2.0</div><div>Record-Route: <sip:10.0.20.102:5062;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes></div><div id="yui_3_16_0_1_1431932745537_77324">Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes></div><div id="yui_3_16_0_1_1431932745537_77326">Via: SIP/2.0/UDP 10.0.20.102:5062;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0</div><div id="yui_3_16_0_1_1431932745537_77329">Via: SIP/2.0/UDP 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060</div><div id="yui_3_16_0_1_1431932745537_77331">From: "301" <<a rel="nofollow" ymailto="mailto:sip%3A301@teste.itcenter.com.pt" target="_blank" href="mailto:sip%3A301@teste.itcenter.com.pt">sip:301@teste.demo.pt</a>>;tag=oztyflbzbx</div><div id="yui_3_16_0_1_1431932745537_77334">To: <<a rel="nofollow" ymailto="mailto:sip%3A401@teste.itcenter.com.pt" target="_blank" href="mailto:sip%3A401@teste.itcenter.com.pt" id="yui_3_16_0_1_1431932745537_77333">sip:401@teste.demo.pt</a>;user=phone></div><div>Call-ID: 3c3a58a25d63-ghfc5xdg1sn0</div><div>CSeq: 1 INVITE</div><div>Max-Forwards: 69</div><div>Contact: <sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1</div><div>X-Serialnumber: 000413262FA0</div><div>P-Key-Flags: resolution="31x13", keys="4"</div><div>User-Agent: snom370/8.4.35</div><div>Accept: application/sdp</div><div>Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE</div><div>Allow-Events: talk, hold, refer, call-info</div><div>Supported: timer, 100rel, replaces, from-change</div><div>Call-Info: <sip:<a rel="nofollow" target="_blank" href="http://teste.itcenter.com.pt/">teste.demo.pt</a>>;appearance-index=1</div><div>Session-Expires: 3600;refresher=uas</div><div>Min-SE: 90</div><div>Content-Type: application/sdp</div><div>Content-Length: 391</div><div>v=0</div><div>o=root 24935823 24935823 IN IP4 192.168.10.147</div><div>s=call</div><div>c=IN IP4 192.168.10.147</div><div>t=0 0</div><div>m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101</div><div>a=rtpmap:0 PCMU/8000.</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:99 G726-32/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div><div>a=rtpmap:4 G723/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=sendrecv</div></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div>INVITE that kamailio send to freeswitch after execute  sdp_remove_codecs_by_id("18"):</div><div><br></div><div><br></div><div><div>INVITE <a rel="nofollow" ymailto="mailto:sip%3A401@teste.demo.pt" target="_blank" href="mailto:sip%3A401@teste.demo.pt">sip:401@teste.demo.pt</a>;user=phone SIP/2.0.</div><div>Record-Route: <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>.</div><div>Record-Route: <sip:10.0.20.102:5062;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>.</div><div>Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>.</div><div>Via: SIP/2.0/UDP 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0.</div><div>Via: SIP/2.0/UDP 10.0.20.102:5062;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0.</div><div>Via: SIP/2.0/UDP 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060.</div><div>From: "301" <<a rel="nofollow" ymailto="mailto:sip%3A301@teste.demo.pt" target="_blank" href="mailto:sip%3A301@teste.demo.pt">sip:301@teste.demo.pt</a>>;tag=zvjgcz9zs9.</div><div>To: <<a rel="nofollow" ymailto="mailto:sip%3A401@teste.demo.pt" target="_blank" href="mailto:sip%3A401@teste.demo.pt">sip:401@teste.demo.pt</a>;user=phone>.</div><div>Call-ID: 3c3a7c84e065-pr2hm0uk9yfz.</div><div>CSeq: 2 INVITE.</div><div>Max-Forwards: 68.</div><div>Contact: <sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1.</div><div>X-Serialnumber: 000413262FA0.</div><div>P-Key-Flags: resolution="31x13", keys="4".</div><div>User-Agent: snom370/<a rel="nofollow" target="_blank" onclick="return theMainWindow.showLinkWarning(this)" href="http://8.4.0.35/">8.4.35.</a></div><div>Accept: application/sdp.</div><div>Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE.</div><div>Allow-Events: talk, hold, refer, call-info.</div><div>Supported: timer, 100rel, replaces, from-change.</div><div>Call-Info: <sip:<a rel="nofollow" target="_blank" href="http://teste.itcenter.com.pt/">teste.itcenter.com.pt</a>>;appearance-index=1.</div><div>Session-Expires: 3600;refresher=uas.</div><div>Min-SE: 90.</div><div>Content-Type: application/sdp.</div><div>Content-Length: 403.</div><div>.<br></div><div>v=0.</div><div>o=root <a rel="nofollow" href="">228603317</a> <a rel="nofollow" href="">228603317</a> IN IP4 100.64.250.4.</div><div>s=call.</div><div>c=IN IP4 100.64.250.4.</div><div>t=0 0.</div><div>m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101.</div><div>a=rtpmap:0 PCMU/8000.</div><div>a=rtpmap:8 PCMA/8000.</div><div>a=rtpmap:9 G722/8000.</div><div>a=rtpmap:99 G726-32/8000.</div><div>a=rtpmap:3 GSM/8000.</div><div>a=rtpmap:18 G729/8000.</div><div>a=fmtp:18 annexb=no.</div><div>a=rtpmap:4 G723/8000.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=ptime:20.</div><div>a=sendrecv.</div><div>a=rtcp:49405.</div></div><div><br></div><div><br></div><div>SDP body has no changes related with codecs.</div><div><br></div><div><br></div><div>Anyone call help please.</div><div><br></div>Thank you</div><div>BR</div><div>José Seabra<br clear="all"><div>-- <br></div><div>Cumprimentos<div>José Seabra</div></div>
</div></div></div><br>_______________________________________________<br>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br><a ymailto="mailto:sr-users@lists.sip-router.org" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br><br><br></div> </div> </div>  </div></body></html>