[SR-Users] Function sdp_remove_codecs_by_id seems to be not working

José Seabra joseseabra4 at gmail.com
Tue May 19 15:13:56 CEST 2015


Hello,

I added the following xlog before function

*xlog("L_INFO", "[SDPOPS] executing function
sdp_remove_codecs_by_id($avp(s:codecs_to_remove)) ID=$ci\n");*

and if i have set debug=2 I can see the xlog message, if I change it to 3 I
cannot see my message, even if I set xlog with L_DBG i cannot see the
message in syslog, this is a weird behavior, can be something wrong with my
rsyslog service?

I did a test that was edit the c function sdp_remove_codecs_by_id in
sdpops_mod.c and i changed the log message from LM_DBG to LM_INFO, then i
compiled and ran again the test and i can see the internal log messages
from function sdp_remove_codecs_by_id.

syslog:

May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Call
flow id '18' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Call
flow name 'queue test' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Action
Type 'CallQueue' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Has
object left '0' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Call
Queues - R=sip:400 at test.centrex.coditel.be;user=phone
 ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>:
Relaying request to freeSWITCH, M=INVITE, du='sip:10.0.20.26:5060',F=
sip:201 at test.centrex.coditel.be - R=sip:400 at test.centrex.coditel.be;user=phone
ID=3134333230343038333536373236-i0wa7mreng1w
*May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>:
[SDPOPS] executing function sdp_remove_codecs_by_id(18)
ID=3134333230343038333536373236-i0wa7mreng1w*
*May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: sdpops
[sdpops_mod.c:331]: sdp_remove_codecs_by_id(): attempting to remove codecs
from sdp: [18]*



BR
José Seabra

2015-05-19 12:25 GMT+01:00 Daniel-Constantin Mierla <miconda at gmail.com>:

>  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>:
>
>>  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 <sip%3A401 at teste.itcenter.com.pt>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 <sip%3A301 at teste.itcenter.com.pt>
>> >;tag=oztyflbzbx
>> To: <sip:401 at teste.demo.pt <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;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>;tag=zvjgcz9zs9.
>> To: <sip:401 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.
>> 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>;appearance-index=1.
>> Session-Expires: 3600;refresher=uas.
>> Min-SE: 90.
>> Content-Type: application/sdp.
>> Content-Length: 403.
>> .
>>  v=0.
>> o=root 228603317 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
>>
>>
>>  _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/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
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
>  --
> Cumprimentos
> José Seabra
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio World Conference, May 27-29, 2015
> Berlin, Germany - http://www.kamailioworld.com
>
>


-- 
Cumprimentos
José Seabra
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