Hi there
I have the following problem - I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage().
Originating INVITE with the "clearmode"
m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000
I do the following in the config route[SDP] { xlog(, "L_INFO", "SDP: request method $rm");
# remove CLEARMODE if Colt if($avp(s:todirection) =~"^MyCarrier+") {
if (is_method("INVITE")) { xlog(, "L_INFO", "SDP remove: request method $rm"); msg_apply_changes(); sdp_remove_codecs_by_name("CLEARMODE"); #sdp_remove_codecs_by_id("125");
} } } This block gets hit as I can see with the xlog entry.
In the sent INVITE the "Clearmode" is still in the sdp header
a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000
So with rtpproxy - it worked With rtpengine not any more ...
Any idea?
KR, Oli
Hi Oli,
i am happy not beeing the only one with trouble with Colt... ;-) The question is, when do you call rtpengine_manage? After this block or before? A major difference between RTPEngine and RTPProxy is, that RTPProxy only replaces IP/Port, while RTPEngine takes the complete SDP and returns a new SDP. Probably, you want to sdp_remove_codecs(), call msg_apply_changes() and then send it to RTPEngine. Or send it to RTPEngine, msg_apply_changes() and then sdp_remove_codecs().
What you are trying to do works: We are using exact the same mechanism to remove AMR Octet aligned variants from the SDP (you find it in the examples folder for the Proxy-CSCF). We remove the codec and then send it to RTPEngine, but also the other direction should work.
Thanks, Carsten
2016-11-16 10:03 GMT+01:00 Oliver Roth oliver.roth@nexphone.ch:
Hello,
are you executing rtpengine_manage() before or after removing the codec?
Cheers, Daniel
On 16/11/16 10:03, Oliver Roth wrote:
Hi,
I guess it is after executing rtpengine_maange() – but not really sure. How can I check that?
I get a different sdp header that is going out of the gateway
v=0 o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38 s=Dialogic-SIP c=IN IP4 185.49.222.198 t=0 0 m=audio 20306 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=sendrecv a=rtcp:20307 a=ice-ufrag:UMTBynMy a=ice-pwd:63JdnvLS7cMyCQ978BA6syPmiI a=candidate:GUTMVOhP7VJyBkZg 1 UDP 2130706431 185.49.222.198 20306 typ host a=candidate:GUTMVOhP7VJyBkZg 2 UDP 2130706430 185.49.222.198 20307 typ host
Incoming was
v=0 o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38 s=Dialogic-SIP c=IN IP4 213.173.185.39 t=0 0 m=audio 9036 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - -
Von: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla Gesendet: Mittwoch, 16. November 2016 10:34 An: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked
Hello,
are you executing rtpengine_manage() before or after removing the codec?
Cheers, Daniel
On 16/11/16 10:03, Oliver Roth wrote: Hi there
I have the following problem – I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage().
Originating INVITE with the “clearmode”
m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000
I do the following in the config route[SDP] { xlog(, "L_INFO", "SDP: request method $rm");
# remove CLEARMODE if Colt if($avp(s:todirection) =~"^MyCarrier+") {
if (is_method("INVITE")) { xlog(, "L_INFO", "SDP remove: request method $rm"); msg_apply_changes(); sdp_remove_codecs_by_name("CLEARMODE"); #sdp_remove_codecs_by_id("125");
} } } This block gets hit as I can see with the xlog entry.
In the sent INVITE the “Clearmode” is still in the sdp header
a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000
So with rtpproxy – it worked With rtpengine not any more …
Any idea?
KR, Oli
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sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
Hello,
On 16/11/16 14:59, Oliver Roth wrote:
load debugger module and enable cfgtrace option via modparam. Then you should see what functions are executed from config.
Cheers, Daniel
Hi,
The codec is removed before sending it to rtpengine … See the log below
Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: *** cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/kamailio-gw.cfg] l=1078 a=2 n=return Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: sdpops [sdpops_mod.c:199]: sdp_remove_str_codec_id_attrs(): removing line: a=rtpmap:125 CLEARMODE/8000 Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: *** cfgtrace:branch_route=[MANAGE_BRANCH] c=[/etc/kamailio/kamailio-gw.cfg] l=1889 a=24 n=rtpengine_manage Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: <core> [mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7fe376c30010, 536) called from rtpengine: bencode.c: __bencode_piece_new(79) Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: rtpengine [rtpengine_funcs.c:140]: check_content_type(): type <application/sdp> found valid
But in the outgoing sdp the codec is still listed
v=0 o=Dialogic_SDP 4043679 0 IN IP4 213.173.185.46 s=Dialogic-SIP c=IN IP4 213.173.185.47 t=0 0 m=audio 9008 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - -
Any ideas?
Kr, Oli
Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Gesendet: Freitag, 18. November 2016 09:27 An: Oliver Roth oliver.roth@nexphone.ch; Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Betreff: Re: AW: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked
Hello,
On 16/11/16 14:59, Oliver Roth wrote: Hi,
I guess it is after executing rtpengine_maange() – but not really sure. How can I check that? load debugger module and enable cfgtrace option via modparam. Then you should see what functions are executed from config.
Cheers, Daniel
I get a different sdp header that is going out of the gateway
v=0 o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38 s=Dialogic-SIP c=IN IP4 185.49.222.198 t=0 0 m=audio 20306 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=sendrecv a=rtcp:20307 a=ice-ufrag:UMTBynMy a=ice-pwd:63JdnvLS7cMyCQ978BA6syPmiI a=candidate:GUTMVOhP7VJyBkZg 1 UDP 2130706431 185.49.222.198 20306 typ host a=candidate:GUTMVOhP7VJyBkZg 2 UDP 2130706430 185.49.222.198 20307 typ host
Incoming was
v=0 o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38 s=Dialogic-SIP c=IN IP4 213.173.185.39 t=0 0 m=audio 9036 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - -
Von: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla Gesendet: Mittwoch, 16. November 2016 10:34 An: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked
Hello,
are you executing rtpengine_manage() before or after removing the codec?
Cheers, Daniel
On 16/11/16 10:03, Oliver Roth wrote: Hi there
I have the following problem – I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage().
Originating INVITE with the “clearmode”
m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000
I do the following in the config route[SDP] { xlog(, "L_INFO", "SDP: request method $rm");
# remove CLEARMODE if Colt if($avp(s:todirection) =~"^MyCarrier+") {
if (is_method("INVITE")) { xlog(, "L_INFO", "SDP remove: request method $rm"); msg_apply_changes(); sdp_remove_codecs_by_name("CLEARMODE"); #sdp_remove_codecs_by_id("125");
} } } This block gets hit as I can see with the xlog entry.
In the sent INVITE the “Clearmode” is still in the sdp header
a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000
So with rtpproxy – it worked With rtpengine not any more …
Any idea?
KR, Oli
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/micondahttp://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
Hi Oli,
if you remove the codec before sending it to RTPEngine, you should call msg_apply_changes() after removing the codec.
Thanks, Carsten
2016-11-18 9:39 GMT+01:00 Oliver Roth oliver.roth@nexphone.ch:
Found the problem with msg_apply_changes: cannot apply msg changes after adding record-route header
see log below:
nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4456]: ERROR: *** cfgtrace:request_route=[SDP] c=[/etc/kamailio/kamailio-gw.cfg] l=745 a=24 n=msg_apply_changes x NoxNov 18 10:04:27 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4456]: ERROR: textopsx [textopsx.c:171]: msg_apply_changes_f(): cannot apply msg changes after adding record-route header - it breaks conditional 2nd header
Any idea?
-----Ursprüngliche Nachricht----- Von: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Carsten Bock Gesendet: Freitag, 18. November 2016 09:42 An: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked
Hi Oli,
if you remove the codec before sending it to RTPEngine, you should call msg_apply_changes() after removing the codec.
Thanks, Carsten
2016-11-18 9:39 GMT+01:00 Oliver Roth oliver.roth@nexphone.ch:
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Move the record_route() function to be executed somewhere after the msg_apply_changes().
Cheers, Daniel
On 18/11/16 10:10, Oliver Roth wrote:
That helped! Now it works.
Thanks Daniel!
KR, Oli
-----Ursprüngliche Nachricht----- Von: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla Gesendet: Freitag, 18. November 2016 15:21 An: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Betreff: ***SPAM***Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked
Move the record_route() function to be executed somewhere after the msg_apply_changes().
Cheers, Daniel
On 18/11/16 10:10, Oliver Roth wrote:
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Great, welcome!
Daniel
On 23/11/2016 10:05, Oliver Roth wrote:
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Carsten Bock -
Daniel-Constantin Mierla -
Oliver Roth