SDP Codec not removed with RTPengine - but with rtpproxy it worked - sr-users

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SDP Codec not removed with RTPengine - but with rtpproxy it worked

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Oliver Roth

16 Nov 2016 16 Nov '16

9:03 a.m.

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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Carsten Bock

16 Nov 16 Nov

9:32 a.m.

New subject: SDP Codec not removed with RTPengine - but with rtpproxy it worked

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:

...

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.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

Daniel-Constantin Mierla

9:34 a.m.

New subject: 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.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 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

Oliver Roth

1:59 p.m.

New subject: SDP Codec not removed with RTPengine - but with rtpproxy it worked

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

_______________________________________________

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/#!/miconda - http://www.linkedin.com/in/miconda

Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

Daniel-Constantin Mierla

18 Nov 18 Nov

8:27 a.m.

New subject: 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.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.org <mailto:sr-users@lists.sip-router.org>

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda http://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

Oliver Roth

8:39 a.m.

New subject: SDP Codec not removed with RTPengine - but with rtpproxy it worked

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

Incoming was

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 mailto: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/#!/miconda http://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

Carsten Bock

8:42 a.m.

New subject: 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:

...

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.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.org

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

--

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

--

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

Oliver Roth

9:10 a.m.

New subject: SDP Codec not removed with RTPengine - but with rtpproxy it worked

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:

...

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.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.org

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

--

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

--

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

-- Carsten Bock CEO (Geschäftsführer)

ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany

http://www.ng-voice.com mailto:carsten@ng-voice.com

Office +49 40 5247593-40 Fax +49 40 5247593-99

Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

_______________________________________________ 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

Daniel-Constantin Mierla

2:20 p.m.

New subject: 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:

...

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:

...
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.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.org

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

--

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

--

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
-- Carsten Bock CEO (Geschäftsführer)

ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany

http://www.ng-voice.com mailto:carsten@ng-voice.com

Office +49 40 5247593-40 Fax +49 40 5247593-99

Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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 _______________________________________________ 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

-- 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

Oliver Roth

23 Nov 23 Nov

9:05 a.m.

New subject: ***SPAM***Re: SDP Codec not removed with RTPengine - but with rtpproxy it worked

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:

...

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:

...
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.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.org

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

--

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

--

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
-- Carsten Bock CEO (Geschäftsführer)

ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany

http://www.ng-voice.com mailto:carsten@ng-voice.com

Office +49 40 5247593-40 Fax +49 40 5247593-99

Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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 _______________________________________________ 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

-- 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

Daniel-Constantin Mierla

1:36 p.m.

New subject: ***SPAM***Re: SDP Codec not removed with RTPengine - but with rtpproxy it worked

Great, welcome!

Daniel

On 23/11/2016 10:05, 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:

...
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:

...
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.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.org

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

--

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

--

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
-- Carsten Bock CEO (Geschäftsführer)

ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany

http://www.ng-voice.com mailto:carsten@ng-voice.com

Office +49 40 5247593-40 Fax +49 40 5247593-99

Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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 _______________________________________________ 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
-- 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 _______________________________________________ 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

-- 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

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Carsten Bock
Daniel-Constantin Mierla
Oliver Roth