[SR-Users] Function sdp_remove_codecs_by_id seems to be not working
Daniel-Constantin Mierla
miconda at gmail.com
Mon May 18 14:26:42 CEST 2015
Hello,
can you run with debug=3 and see if the function is actually executed?
Cheers,
Daniel
On 18/05/15 12:31, José Seabra wrote:
> Hello,
>
> 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.
>
> I'm using this function in request route.
>
>
> Kamailio version is 4.2.2.
>
>
> INVITE that kamailio receives from phone:
>
> INVITE sip:401 at teste.d <mailto:sip%3A401 at teste.itcenter.com.pt>emo.pt
> <http://emo.pt>;user=phone SIP/2.0
> Record-Route:
> <sip:10.0.20.102:5062;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes>
> Record-Route:
> <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes>
> Via: SIP/2.0/UDP
> 10.0.20.102:5062;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0
> Via: SIP/2.0/UDP
> 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060
> From: "301" <sip:301 at teste.demo.pt
> <mailto:sip%3A301 at teste.itcenter.com.pt>>;tag=oztyflbzbx
> To: <sip:401 at teste.demo.pt
> <mailto:sip%3A401 at teste.itcenter.com.pt>;user=phone>
> Call-ID: 3c3a58a25d63-ghfc5xdg1sn0
> CSeq: 1 INVITE
> Max-Forwards: 69
> Contact:
> <sip:301 at 192.168.10.147:5060;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1
> X-Serialnumber: 000413262FA0
> P-Key-Flags: resolution="31x13", keys="4"
> User-Agent: snom370/8.4.35
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Call-Info: <sip:teste.demo.pt
> <http://teste.itcenter.com.pt>>;appearance-index=1
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 391
> v=0
> o=root 24935823 24935823 IN IP4 192.168.10.147
> s=call
> c=IN IP4 192.168.10.147
> t=0 0
> m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
>
>
>
>
> INVITE that kamailio send to freeswitch after execute
> sdp_remove_codecs_by_id("18"):
>
>
> INVITE sip:401 at teste.demo.pt
> <mailto:sip%3A401 at teste.demo.pt>;user=phone SIP/2.0.
> Record-Route:
> <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>.
> Record-Route:
> <sip:10.0.20.102:5062;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>.
> Record-Route:
> <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>.
> Via: SIP/2.0/UDP
> 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0.
> Via: SIP/2.0/UDP
> 10.0.20.102:5062;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0.
> Via: SIP/2.0/UDP
> 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060.
> From: "301" <sip:301 at teste.demo.pt
> <mailto:sip%3A301 at teste.demo.pt>>;tag=zvjgcz9zs9.
> To: <sip:401 at teste.demo.pt <mailto:sip%3A401 at teste.demo.pt>;user=phone>.
> Call-ID: 3c3a7c84e065-pr2hm0uk9yfz.
> CSeq: 2 INVITE.
> Max-Forwards: 68.
> Contact:
> <sip:301 at 192.168.10.147:5060;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1.
> X-Serialnumber: 000413262FA0.
> P-Key-Flags: resolution="31x13", keys="4".
> User-Agent: snom370/8.4.35. <http://8.4.35.>
> Accept: application/sdp.
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE.
> Allow-Events: talk, hold, refer, call-info.
> Supported: timer, 100rel, replaces, from-change.
> Call-Info: <sip:teste.itcenter.com.pt
> <http://teste.itcenter.com.pt>>;appearance-index=1.
> Session-Expires: 3600;refresher=uas.
> Min-SE: 90.
> Content-Type: application/sdp.
> Content-Length: 403.
> .
> v=0.
> o=root 228603317 <tel:228603317> 228603317 <tel:228603317> IN IP4
> 100.64.250.4.
> s=call.
> c=IN IP4 100.64.250.4.
> t=0 0.
> m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:9 G722/8000.
> a=rtpmap:99 G726-32/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:4 G723/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:49405.
>
>
> SDP body has no changes related with codecs.
>
>
> Anyone call help please.
>
> Thank you
> BR
> José Seabra
> --
> Cumprimentos
> José Seabra
>
>
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--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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