<div dir="ltr">Hello,<div><br></div><div>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>I'm using this function in request route.</div><div><br></div><div><br></div><div>Kamailio version is 4.2.2.</div><div><br clear="all"><div><br></div><div>INVITE that kamailio receives from phone:</div><div><br></div><div><div>INVITE <a href="mailto:sip%3A401@teste.itcenter.com.pt" target="_blank">sip:401@teste.d</a><a href="http://emo.pt" target="_blank">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>Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes></div><div>Via: SIP/2.0/UDP 10.0.20.102:5062;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0</div><div>Via: SIP/2.0/UDP 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060</div><div>From: "301" <<a href="mailto:sip%3A301@teste.itcenter.com.pt" target="_blank">sip:301@teste.demo.pt</a>>;tag=oztyflbzbx</div><div>To: <<a href="mailto:sip%3A401@teste.itcenter.com.pt" target="_blank">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 href="http://teste.itcenter.com.pt" target="_blank">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 href="mailto:sip%3A401@teste.demo.pt" target="_blank">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 href="mailto:sip%3A301@teste.demo.pt" target="_blank">sip:301@teste.demo.pt</a>>;tag=zvjgcz9zs9.</div><div>To: <<a href="mailto:sip%3A401@teste.demo.pt" target="_blank">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 href="http://8.4.35." target="_blank">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 href="http://teste.itcenter.com.pt" target="_blank">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 href="tel:228603317" value="+351228603317" target="_blank">228603317</a> <a href="tel:228603317" value="+351228603317" target="_blank">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>
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