[SR-Users] Function sdp_remove_codecs_by_id seems to be not working
Daniel-Constantin Mierla
miconda at gmail.com
Tue May 19 13:25:56 CEST 2015
Hello,
can you enable cfgtrace via debugger module or add an xlog just before
calling the function in configuration file and see if related message
appears in syslog?
Cheers,
Daniel
On 19/05/15 11:05, José Seabra wrote:
> Hello,
> Thank you for your reply
>
> I ran kamailio with debug=3 and log_stderror=yes and the only thing
> that i see related with function sdp_remove_codecs_by_id is:
>
> 0(4707) DEBUG: <core> [route.c:907]: fix_actions(): fixing
> sdp_remove_codecs_by_id()
>
>
> if i set debug=3 and log_stderror=no then i look for syslog file
> where kamailio is writting logs, and i don't see anything related with
> function sdp_remove_codecs_by_id.
>
> I'm not using msg_apply_changes function.
>
> Thank you for your support
>
> BR
> José Seabra
>
> 2015-05-18 13:26 GMT+01:00 Daniel-Constantin Mierla <miconda at gmail.com
> <mailto:miconda at gmail.com>>:
>
> 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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>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda
> Kamailio World Conference, May 27-29, 2015
> Berlin, Germany - http://www.kamailioworld.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> list
> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
> --
> Cumprimentos
> José Seabra
--
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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